diff --git a/app/CMakeLists.txt b/app/CMakeLists.txt
index 7adc98e..98a72d9 100644
--- a/app/CMakeLists.txt
+++ b/app/CMakeLists.txt
@@ -44,6 +44,7 @@ add_library( # Sets the name of the library.
             src/main/cpp/metadata/ffmpeg_media_retriever.c
             src/main/cpp/yuv/yuv_converter.cpp
             src/main/cpp/media_transcode.cpp
+            src/main/cpp/audio_resample.cpp
         )
 
 add_library( ffmpeg
diff --git a/app/src/main/cpp/audio_resample.cpp b/app/src/main/cpp/audio_resample.cpp
new file mode 100644
index 0000000..59031c7
--- /dev/null
+++ b/app/src/main/cpp/audio_resample.cpp
@@ -0,0 +1,222 @@
+//
+// Created by xu fulong on 2022/7/12.
+//
+
+#include <jni.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+#include "libavformat/avformat.h"
+#include <libavutil/opt.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/samplefmt.h>
+#include <libswresample/swresample.h>
+#ifdef __cplusplus
+}
+#endif
+
+#include "ffmpeg_jni_define.h"
+
+#define ALOGE(Format, ...) LOGE("audio_resample", Format, ##__VA_ARGS__)
+
+static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
+{
+    *fmt = nullptr;
+
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+            { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+            { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+            { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+            { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+            { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+
+    for (int i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    ALOGE("Sample format %s not supported as output format, msg=%s\n",
+            av_get_sample_fmt_name(sample_fmt), strerror(errno));
+    return AVERROR(EINVAL);
+}
+
+int resampling(const char *src_filename, const char *dst_filename, int dst_rate)
+{
+    int src_rate = 0;
+    int src_linesize;
+    int src_nb_channels;
+    int src_nb_samples = 0;
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO;
+    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_S16;
+
+    int dst_bufsize;
+    int dst_linesize;
+    int dst_nb_channels;
+    uint8_t **dst_data = nullptr;
+    int dst_nb_samples, max_dst_nb_samples;
+    int64_t dst_ch_layout = AV_CH_LAYOUT_SURROUND;
+    enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;
+
+    int ret;
+    const char *fmt;
+    AVPacket packet;
+    AVFrame *frame;
+    int got_frame_ptr;
+    struct SwrContext *swr_ctx;
+    AVFormatContext *iformat_ctx = nullptr;
+    AVFormatContext *oformat_ctx = nullptr;
+
+    ret = avformat_open_input(&iformat_ctx, src_filename, nullptr, nullptr);
+    if (ret < 0) {
+        ALOGE("open input fail, path=%s, ret=%d", src_filename, ret);
+        goto end;
+    }
+
+    /* create resample context */
+    swr_ctx = swr_alloc();
+    if (!swr_ctx) {
+        ALOGE("Could not allocate resample context:%s\n", strerror(errno));
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* set options */
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
+
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
+
+    /* initialize the resampling context */
+    if ((ret = swr_init(swr_ctx)) < 0) {
+        ALOGE("Failed to initialize the resampling context:%s\n", strerror(errno));
+        goto end;
+    }
+
+    /* allocate source and destination samples buffers */
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = dst_nb_samples =
+            (int) av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+
+    /* buffer is going to be directly written to a raw-audio file, no alignment */
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
+    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
+                                             dst_nb_samples, dst_sample_fmt, 0);
+    if (ret < 0) {
+        ALOGE("Could not allocate destination samples:%s\n", strerror(errno));
+        goto end;
+    }
+
+    avformat_alloc_output_context2(&oformat_ctx, nullptr, nullptr, dst_filename);
+    av_dump_format(oformat_ctx, 0, dst_filename, 1);
+    if (!(oformat_ctx->oformat->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&oformat_ctx->pb, dst_filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            ALOGE("Could not open output file %s\n", dst_filename);
+            return ret;
+        }
+    }
+    /* init muxer, write output file header */
+    ret = avformat_write_header(oformat_ctx, nullptr);
+    if (ret < 0) {
+        ALOGE("Error occurred when opening output file\n");
+        return ret;
+    }
+
+    while (av_read_frame(iformat_ctx, &packet) >= 0) {
+        /* compute destination number of samples */
+        dst_nb_samples = (int) av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
+                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_freep(&dst_data[0]);
+            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
+                                   dst_nb_samples, dst_sample_fmt, 1);
+            if (ret < 0)
+                break;
+            max_dst_nb_samples = dst_nb_samples;
+        }
+
+        ret = avcodec_decode_audio4(iformat_ctx, frame, &got_frame_ptr, packet);
+        if (ret < 0) {
+            ALOGE("decode audio error:%d\n", ret);
+            continue;
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples);
+        if (ret < 0) {
+            ALOGE("Error while converting:%s\n", strerror(errno));
+            goto end;
+        }
+        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
+                                                 ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            ALOGE("Could not get sample buffer size:%s\n", strerror(errno));
+            goto end;
+        }
+
+        AVStream *in_stream = iformat_ctx->streams[0];
+        AVStream *out_stream = oformat_ctx->streams[0];
+        pkt->pts = av_rescale_q_rnd(packet.pts, in_stream->time_base, out_stream->time_base,
+                                    static_cast<AVRounding>(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
+        pkt->dts = av_rescale_q_rnd(packet.dts, in_stream->time_base, out_stream->time_base,
+                                    static_cast<AVRounding>(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
+        pkt->duration = av_rescale_q(packet.duration, in_stream->time_base, out_stream->time_base);
+        pkt->pos = -1;
+
+        av_interleaved_write_frame(oformat_ctx, pkt);
+
+        av_packet_unref(&packet);
+    }
+
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
+        goto end;
+    ALOGE("Resampling succeeded. Play the output file with the command:\n"
+                    "ffplay -f %s -channel_layout %" PRId64 " -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
+
+end:
+
+    if (dst_data)
+        av_freep(&dst_data[0]);
+    av_freep(&dst_data);
+
+    swr_free(&swr_ctx);
+
+    av_write_trailer(oformat_ctx);
+    if (!(oformat_ctx->oformat->flags & AVFMT_NOFILE)) {
+        avio_close(oformat_ctx->pb);
+    }
+    avformat_free_context(oformat_ctx);
+    avformat_close_input(&iformat_ctx);
+
+    return ret;
+}
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+VIDEO_PLAYER_FUNC(int, audioResample, jstring srcFile, jstring dstFile, int sampleRate) {
+    const char *src_file = env->GetStringUTFChars(srcFile, JNI_FALSE);
+    const char *dst_file = env->GetStringUTFChars(dstFile, JNI_FALSE);
+    int ret = resampling(src_file, dst_file, sampleRate);
+    env->ReleaseStringUTFChars(dstFile, dst_file);
+    env->ReleaseStringUTFChars(srcFile, src_file);
+    return ret;
+}
+#ifdef __cplusplus
+}
+#endif
\ No newline at end of file
diff --git a/app/src/main/cpp/media_transcode.cpp b/app/src/main/cpp/media_transcode.cpp
index c1574b5..30dd8fc 100644
--- a/app/src/main/cpp/media_transcode.cpp
+++ b/app/src/main/cpp/media_transcode.cpp
@@ -19,7 +19,7 @@ extern "C" {
 
 #include "ffmpeg_jni_define.h"
 
-#define ALOGE(Format, ...) LOGE("Transcde", Format, ##__VA_ARGS__)
+#define ALOGE(Format, ...) LOGE("Transcode", Format, ##__VA_ARGS__)
 
 static AVFormatContext *ifmt_ctx;
 static AVFormatContext *ofmt_ctx;
@@ -597,7 +597,7 @@ end:
     if (ret < 0)
         ALOGE("Error occurred: %s\n", av_err2str(ret));
 
-    return ret ? 1 : 0;
+    return ret;
 }
 
 #ifdef __cplusplus
diff --git a/app/src/main/cpp/video_cutting.cpp b/app/src/main/cpp/video_cutting.cpp
index 8d57f66..6f4ade4 100644
--- a/app/src/main/cpp/video_cutting.cpp
+++ b/app/src/main/cpp/video_cutting.cpp
@@ -80,6 +80,7 @@ AVPacket* CutVideo::copy_packet(AVFormatContext *ifmt_ctx, AVPacket *packet) {
         pkt->dts = av_rescale_q_rnd(pkt->dts, in_stream->time_base, out_stream->time_base,
                                     static_cast<AVRounding>(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
         pkt->duration = av_rescale_q(pkt->duration, in_stream->time_base, out_stream->time_base);
+        pkt->pos = -1;
         return pkt;
     }
     return nullptr;
diff --git a/app/src/main/java/com/frank/ffmpeg/VideoPlayer.java b/app/src/main/java/com/frank/ffmpeg/VideoPlayer.java
index b7a8fbd..d772b5a 100644
--- a/app/src/main/java/com/frank/ffmpeg/VideoPlayer.java
+++ b/app/src/main/java/com/frank/ffmpeg/VideoPlayer.java
@@ -28,6 +28,8 @@ public class VideoPlayer {
 
     public native int executeTranscode(String inputFile, String outputFile);
 
+    public native int audioResample(String outputFile, int sampleRate);
+
     /**
      * Create an AudioTrack instance for JNI calling
      *