swr

If there is exactly one producer, then it can assume that the remaining
space in the buffer will only increase until it write something.

This assumption may allow the producer to write to the buffer (up to a
known safe size) without any synchronization mechanism, thus allowing
to read and write different parts of the buffer in parallel.

The producer can then commit the write with lock held, and update its
knowledge of the safe empty remaining space.

app/meson.build

@@ -31,7 +31,6 @@ src = [
     'src/version.c',
     'src/video_buffer.c',
     'src/util/acksync.c',
-    'src/util/average.c',
     'src/util/bytebuf.c',
     'src/util/file.c',
     'src/util/intmap.c',

app/src/audio_player.c

@@ -4,27 +4,11 @@
 
 #include "util/log.h"
 
-#define SC_AUDIO_PLAYER_NDEBUG // comment to debug
-
-/** Downcast frame_sink to sc_audio_player */
+/** Downcast frame_sink to sc_v4l2_sink */
 #define DOWNCAST(SINK) container_of(SINK, struct sc_audio_player, frame_sink)
 
-#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
-#define SC_SDL_SAMPLE_FMT AUDIO_F32
-
-#define SC_AUDIO_OUTPUT_BUFFER_SAMPLES 480 // 10ms at 48000Hz
-
-// The target number of buffered samples between the producer and the consumer.
-// This value is directly use for compensation.
-#define SC_TARGET_BUFFERED_SAMPLES (3 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES)
-
-// If the consumer is too late, skip samples to keep at most this value
-#define SC_BUFFERED_SAMPLES_THRESHOLD 2400 // 50ms at 48000Hz
-
-// Use a ring-buffer of 1 second (at 48000Hz) between the producer and the
-// consumer. It too big, but it guarantees that the producer and the consumer
-// will be able to access it in parallel without locking.
-#define SC_BYTEBUF_SIZE_IN_SAMPLES 48000
+#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_S16
+#define SC_SDL_SAMPLE_FMT AUDIO_S16
 
 void
 sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
@@ -36,49 +20,21 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
     assert(len_int > 0);
     size_t len = len_int;
 
-#ifndef SC_AUDIO_PLAYER_NDEBUG
-    LOGD("[Audio] SDL callback requests %" SC_PRIsizet " samples",
-         len / (ap->nb_channels * ap->out_bytes_per_sample));
-#endif
-
     size_t read = sc_bytebuf_read_remaining(&ap->buf);
-    size_t max_buffered_bytes = SC_BUFFERED_SAMPLES_THRESHOLD
-                              * ap->nb_channels * ap->out_bytes_per_sample;
-    if (read > max_buffered_bytes + len) {
-        size_t skip = read - (max_buffered_bytes + len);
-#ifndef SC_AUDIO_PLAYER_NDEBUG
-        LOGD("[Audio] Buffered samples threshold exceeded: %" SC_PRIsizet
-             " bytes, skipping %" SC_PRIsizet " bytes", read, skip);
-#endif
-        // After this callback, exactly max_buffered_bytes will remain
-        sc_bytebuf_skip(&ap->buf, skip);
-        read = max_buffered_bytes + len;
-    }
-
-    // Number of buffered samples (may be negative on underflow)
-    float buffered_samples = ((float) read - len_int)
-                           / (ap->nb_channels * ap->out_bytes_per_sample);
-    sc_average_push(&ap->avg_buffered_samples, buffered_samples);
-
     if (read) {
         if (read > len) {
             read = len;
         }
         sc_bytebuf_read(&ap->buf, stream, read);
     }
-
     if (read < len) {
         // Insert silence
-#ifndef SC_AUDIO_PLAYER_NDEBUG
-        LOGD("[Audio] Buffer underflow, inserting silence: %" SC_PRIsizet
-             " bytes", len - read);
-#endif
         memset(stream + read, 0, len - read);
     }
 }
 
 static size_t
-sc_audio_player_get_buf_size(struct sc_audio_player *ap, size_t samples) {
+sc_audio_player_get_swr_buf_size(struct sc_audio_player *ap, size_t samples) {
     assert(ap->nb_channels);
     assert(ap->out_bytes_per_sample);
     return samples * ap->nb_channels * ap->out_bytes_per_sample;
@@ -86,7 +42,7 @@ sc_audio_player_get_buf_size(struct sc_audio_player *ap, size_t samples) {
 
 static uint8_t *
 sc_audio_player_get_swr_buf(struct sc_audio_player *ap, size_t min_samples) {
-    size_t min_buf_size = sc_audio_player_get_buf_size(ap, min_samples);
+    size_t min_buf_size = sc_audio_player_get_swr_buf_size(ap, min_samples);
     if (min_buf_size < ap->swr_buf_alloc_size) {
         size_t new_size = min_buf_size + 4096;
         uint8_t *buf = realloc(ap->swr_buf, new_size);
@@ -107,28 +63,8 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
                                 const AVCodecContext *ctx) {
     struct sc_audio_player *ap = DOWNCAST(sink);
 
-    SDL_AudioSpec desired = {
-        .freq = ctx->sample_rate,
-        .format = SC_SDL_SAMPLE_FMT,
-        .channels = ctx->ch_layout.nb_channels,
-        .samples = SC_AUDIO_OUTPUT_BUFFER_SAMPLES,
-        .callback = sc_audio_player_sdl_callback,
-        .userdata = ap,
-    };
-    SDL_AudioSpec obtained;
-
-    ap->device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
-    if (!ap->device) {
-        LOGE("Could not open audio device: %s", SDL_GetError());
-        return false;
-    }
-
-    SwrContext *swr_ctx = swr_alloc();
-    if (!swr_ctx) {
-        LOG_OOM();
-        goto error_close_audio_device;
-    }
-    ap->swr_ctx = swr_ctx;
+    SwrContext *swr_ctx = ap->swr_ctx;
+    assert(swr_ctx);
 
     assert(ctx->sample_rate > 0);
     assert(ctx->ch_layout.nb_channels > 0);
@@ -147,46 +83,39 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
     int ret = swr_init(swr_ctx);
     if (ret) {
         LOGE("Failed to initialize the resampling context");
-        goto error_free_swr_ctx;
+        return false;
     }
 
     ap->sample_rate = ctx->sample_rate;
     ap->nb_channels = ctx->ch_layout.nb_channels;
     ap->out_bytes_per_sample = out_bytes_per_sample;
 
-    size_t bytebuf_size =
-        sc_audio_player_get_buf_size(ap, SC_BYTEBUF_SIZE_IN_SAMPLES);
-
-    bool ok = sc_bytebuf_init(&ap->buf, bytebuf_size);
-    if (!ok) {
-        goto error_free_swr_ctx;
-    }
-
-    ap->safe_empty_buffer = sc_bytebuf_write_remaining(&ap->buf);
-
-    size_t initial_swr_buf_size = sc_audio_player_get_buf_size(ap, 4096);
+    size_t initial_swr_buf_size = sc_audio_player_get_swr_buf_size(ap, 4096);
     ap->swr_buf = malloc(initial_swr_buf_size);
     if (!ap->swr_buf) {
         LOG_OOM();
-        goto error_destroy_bytebuf;
+        return false;
     }
-    ap->swr_buf_alloc_size = initial_swr_buf_size;
 
-    sc_average_init(&ap->avg_buffered_samples, 32);
-    ap->samples_since_resync = 0;
+    SDL_AudioSpec desired = {
+        .freq = ctx->sample_rate,
+        .format = SC_SDL_SAMPLE_FMT,
+        .channels = ctx->ch_layout.nb_channels,
+        .samples = 512, // ~10ms at 48000Hz
+        .callback = sc_audio_player_sdl_callback,
+        .userdata = ap,
+    };
+    SDL_AudioSpec obtained;
+
+    ap->device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
+    if (!ap->device) {
+        LOGE("Could not open audio device: %s", SDL_GetError());
+        return false;
+    }
 
     SDL_PauseAudioDevice(ap->device, 0);
 
     return true;
-
-error_destroy_bytebuf:
-    sc_bytebuf_destroy(&ap->buf);
-error_free_swr_ctx:
-    swr_free(&ap->swr_ctx);
-error_close_audio_device:
-    SDL_CloseAudioDevice(ap->device);
-
-    return false;
 }
 
 static void
@@ -196,10 +125,6 @@ sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
     assert(ap->device);
     SDL_PauseAudioDevice(ap->device, 1);
     SDL_CloseAudioDevice(ap->device);
-
-    free(ap->swr_buf);
-    sc_bytebuf_destroy(&ap->buf);
-    swr_free(&ap->swr_ctx);
 }
 
 static bool
@@ -223,12 +148,12 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink, const AVFrame *frame
         LOGE("Resampling failed: %d", ret);
         return false;
     }
+    LOGI("ret=%d dst_nb_samples=%d\n", ret, dst_nb_samples);
 
-    size_t samples_written = ret;
-    size_t swr_buf_size = sc_audio_player_get_buf_size(ap, samples_written);
-#ifndef SC_AUDIO_PLAYER_NDEBUG
-    LOGI("[Audio] %" SC_PRIsizet " samples written to buffer", samples_written);
-#endif
+    size_t swr_buf_size = sc_audio_player_get_swr_buf_size(ap, ret);
+    LOGI("== swr_buf_size %lu", swr_buf_size);
+
+    // TODO clock drift compensation
 
     // It should almost always be possible to write without lock
     bool can_write_without_lock = swr_buf_size <= ap->safe_empty_buffer;
@@ -245,39 +170,36 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink, const AVFrame *frame
 
     // The next time, it will remain at least the current empty space
     ap->safe_empty_buffer = sc_bytebuf_write_remaining(&ap->buf);
-
-    // Read the value written by the SDL thread under lock
-    float avg;
-    bool has_avg = sc_average_get(&ap->avg_buffered_samples, &avg);
-
     SDL_UnlockAudioDevice(ap->device);
 
-    if (has_avg) {
-        ap->samples_since_resync += samples_written;
-        if (ap->samples_since_resync >= ap->sample_rate) {
-            // Resync every second
-            ap->samples_since_resync = 0;
-
-            int diff = SC_TARGET_BUFFERED_SAMPLES - avg;
-#ifndef SC_AUDIO_PLAYER_NDEBUG
-            LOGI("[Audio] Average buffered samples = %f, compensation %d",
-                 avg, diff);
-#endif
-            // Compensate the diff over 3 seconds (but will be recomputed after
-            // 1 second)
-            int ret = swr_set_compensation(swr_ctx, diff, 3 * ap->sample_rate);
-            if (ret < 0) {
-                LOGW("Resampling compensation failed: %d", ret);
-                // not fatal
-            }
-        }
-    }
-
     return true;
 }
 
-void
-sc_audio_player_init(struct sc_audio_player *ap) {
+bool
+sc_audio_player_init(struct sc_audio_player *ap,
+                     const struct sc_audio_player_callbacks *cbs,
+                     void *cbs_userdata) {
+    bool ok = sc_bytebuf_init(&ap->buf, 128 * 1024);
+    if (!ok) {
+        return false;
+    }
+
+    ap->swr_ctx = swr_alloc();
+    if (!ap->swr_ctx) {
+        sc_bytebuf_destroy(&ap->buf);
+        LOG_OOM();
+        return false;
+    }
+
+    ap->safe_empty_buffer = sc_bytebuf_write_remaining(&ap->buf);
+
+    ap->swr_buf = NULL;
+    ap->swr_buf_alloc_size = 0;
+
+    assert(cbs && cbs->on_ended);
+    ap->cbs = cbs;
+    ap->cbs_userdata = cbs_userdata;
+
     static const struct sc_frame_sink_ops ops = {
         .open = sc_audio_player_frame_sink_open,
         .close = sc_audio_player_frame_sink_close,
@@ -285,4 +207,12 @@ sc_audio_player_init(struct sc_audio_player *ap) {
     };
 
     ap->frame_sink.ops = &ops;
+    return true;
+}
+
+void
+sc_audio_player_destroy(struct sc_audio_player *ap) {
+    sc_bytebuf_destroy(&ap->buf);
+    swr_free(&ap->swr_ctx);
+    free(ap->swr_buf);
 }

app/src/audio_player.h

@@ -5,7 +5,6 @@
 
 #include <stdbool.h>
 #include "trait/frame_sink.h"
-#include <util/average.h>
 #include <util/bytebuf.h>
 #include <util/thread.h>
 
@@ -36,10 +35,6 @@ struct sc_audio_player {
     uint8_t *swr_buf;
     size_t swr_buf_alloc_size;
 
-    // Number of buffered samples (may be negative on underflow)
-    struct sc_average avg_buffered_samples;
-    unsigned samples_since_resync;
-
     const struct sc_audio_player_callbacks *cbs;
     void *cbs_userdata;
 };
@@ -48,7 +43,12 @@ struct sc_audio_player_callbacks {
     void (*on_ended)(struct sc_audio_player *ap, bool success, void *userdata);
 };
 
+bool
+sc_audio_player_init(struct sc_audio_player *ap,
+                     const struct sc_audio_player_callbacks *cbs,
+                     void *cbs_userdata);
+
 void
-sc_audio_player_init(struct sc_audio_player *ap);
+sc_audio_player_destroy(struct sc_audio_player *ap);
 
 #endif

app/src/scrcpy.c

@@ -217,6 +217,17 @@ sc_recorder_on_ended(struct sc_recorder *recorder, bool success,
     }
 }
 
+static void
+sc_audio_player_on_ended(struct sc_audio_player *ap, bool success,
+                         void *userdata) {
+    (void) ap;
+    (void) userdata;
+
+    if (!success) {
+        // TODO
+    }
+}
+
 static void
 sc_video_demuxer_on_ended(struct sc_demuxer *demuxer, bool eos,
                           void *userdata) {
@@ -303,6 +314,7 @@ scrcpy(struct scrcpy_options *options) {
     bool file_pusher_initialized = false;
     bool recorder_initialized = false;
     bool recorder_started = false;
+    bool audio_player_initialized = false;
 #ifdef HAVE_V4L2
     bool v4l2_sink_initialized = false;
 #endif
@@ -674,7 +686,15 @@ aoa_hid_end:
         sc_decoder_add_sink(&s->video_decoder, &s->screen.frame_sink);
 
         if (options->audio) {
-            sc_audio_player_init(&s->audio_player);
+            static const struct sc_audio_player_callbacks audio_player_cbs = {
+                .on_ended = sc_audio_player_on_ended,
+            };
+            if (!sc_audio_player_init(&s->audio_player,
+                                      &audio_player_cbs, NULL)) {
+                goto end;
+            }
+            audio_player_initialized = true;
+
             sc_decoder_add_sink(&s->audio_decoder, &s->audio_player.frame_sink);
         }
     }
@@ -797,6 +817,10 @@ end:
         sc_recorder_destroy(&s->recorder);
     }
 
+    if (audio_player_initialized) {
+        sc_audio_player_destroy(&s->audio_player);
+    }
+
     if (file_pusher_initialized) {
         sc_file_pusher_join(&s->file_pusher);
         sc_file_pusher_destroy(&s->file_pusher);

app/src/v4l2_sink.c

@@ -338,9 +338,9 @@ sc_v4l2_sink_push(struct sc_v4l2_sink *vs, const AVFrame *frame) {
 }
 
 static bool
-sc_v4l2_frame_sink_open(struct sc_frame_sink *sink, const AVCodecContext *ctx) {
+sc_v4l2_frame_sink_open(struct sc_frame_sink *sink) {
     struct sc_v4l2_sink *vs = DOWNCAST(sink);
-    return sc_v4l2_sink_open(vs, ctx);
+    return sc_v4l2_sink_open(vs);
 }
 
 static void

server/src/main/java/com/genymobile/scrcpy/AudioEncoder.java

@@ -1,11 +1,7 @@
 package com.genymobile.scrcpy;
 
-import com.genymobile.scrcpy.wrappers.ServiceManager;
-
 import android.annotation.SuppressLint;
 import android.annotation.TargetApi;
-import android.content.ComponentName;
-import android.content.Intent;
 import android.media.AudioFormat;
 import android.media.AudioRecord;
 import android.media.AudioTimestamp;
@@ -16,7 +12,6 @@ import android.os.Build;
 import android.os.Handler;
 import android.os.HandlerThread;
 import android.os.Looper;
-import android.os.SystemClock;
 
 import java.io.IOException;
 import java.nio.ByteBuffer;
@@ -45,13 +40,10 @@ public final class AudioEncoder {
     }
 
     private static final int SAMPLE_RATE = 48000;
-    private static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_STEREO;
     private static final int CHANNELS = 2;
-    private static final int FORMAT = AudioFormat.ENCODING_PCM_16BIT;
-    private static final int BYTES_PER_SAMPLE = 2;
 
-    private static final int BUFFER_MS = 5; // milliseconds
-    private static final int BUFFER_SIZE = SAMPLE_RATE * CHANNELS * BYTES_PER_SAMPLE * BUFFER_MS / 1000;
+    private static final int BUFFER_MS = 10; // milliseconds
+    private static final int BUFFER_SIZE = SAMPLE_RATE * CHANNELS * BUFFER_MS / 1000;
 
     private final Streamer streamer;
     private final int bitRate;
@@ -80,9 +72,9 @@ public final class AudioEncoder {
 
     private static AudioFormat createAudioFormat() {
         AudioFormat.Builder builder = new AudioFormat.Builder();
-        builder.setEncoding(FORMAT);
+        builder.setEncoding(AudioFormat.ENCODING_PCM_16BIT);
         builder.setSampleRate(SAMPLE_RATE);
-        builder.setChannelMask(CHANNEL_CONFIG);
+        builder.setChannelMask(CHANNELS == 2 ? AudioFormat.CHANNEL_IN_STEREO : AudioFormat.CHANNEL_IN_MONO);
         return builder.build();
     }
 
@@ -96,8 +88,7 @@ public final class AudioEncoder {
         }
         builder.setAudioSource(MediaRecorder.AudioSource.REMOTE_SUBMIX);
         builder.setAudioFormat(createAudioFormat());
-        int minBufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_CONFIG, FORMAT);
-        builder.setBufferSizeInBytes(minBufferSize);
+        builder.setBufferSizeInBytes(1024 * 1024);
         return builder.build();
     }
 
@@ -220,32 +211,6 @@ public final class AudioEncoder {
         }
     }
 
-    private static void startWorkaroundAndroid11() {
-        if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
-            // Android 11 requires Apps to be at foreground to record audio.
-            // Normally, each App has its own user ID, so Android checks whether the requesting App has the user ID that's at the foreground.
-            // But Scrcpy server is NOT an App, it's a Java application started from Android shell, so it has the same user ID (2000) with Android
-            // shell ("com.android.shell").
-            // If there is an Activity from Android shell running at foreground, then the permission system will believe Scrcpy is also in the
-            // foreground.
-            if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
-                Intent intent = new Intent(Intent.ACTION_MAIN);
-                intent.addFlags(Intent.FLAG_ACTIVITY_NEW_TASK);
-                intent.addCategory(Intent.CATEGORY_LAUNCHER);
-                intent.setComponent(new ComponentName(FakeContext.PACKAGE_NAME, "com.android.shell.HeapDumpActivity"));
-                ServiceManager.getActivityManager().startActivityAsUserWithFeature(intent);
-                // Wait for activity to start
-                SystemClock.sleep(150);
-            }
-        }
-    }
-
-    private static void stopWorkaroundAndroid11() {
-        if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
-            ServiceManager.getActivityManager().forceStopPackage(FakeContext.PACKAGE_NAME);
-        }
-    }
-
     @TargetApi(Build.VERSION_CODES.M)
     public void encode() throws IOException {
         if (Build.VERSION.SDK_INT < Build.VERSION_CODES.R) {
@@ -263,6 +228,7 @@ public final class AudioEncoder {
         try {
             Codec codec = streamer.getCodec();
             mediaCodec = createMediaCodec(codec, encoderName);
+            recorder = createAudioRecord();
 
             mediaCodecThread = new HandlerThread("AudioEncoder");
             mediaCodecThread.start();
@@ -271,19 +237,7 @@ public final class AudioEncoder {
             mediaCodec.setCallback(new EncoderCallback(), new Handler(mediaCodecThread.getLooper()));
             mediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
 
-            startWorkaroundAndroid11();
-            try {
-                recorder = createAudioRecord();
-                recorder.startRecording();
-            } catch (UnsupportedOperationException e) {
-                if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
-                    Ln.e("Failed to start audio capture");
-                    Ln.e("On Android 11, it is only possible to capture in foreground, make sure that the device is unlocked when starting scrcpy.");
-                    throw new ConfigurationException("Unsupported audio capture");
-                }
-            } finally {
-                stopWorkaroundAndroid11();
-            }
+            recorder.startRecording();
             recorderStarted = true;
 
             final MediaCodec mediaCodecRef = mediaCodec;

server/src/main/java/com/genymobile/scrcpy/wrappers/ActivityManager.java

@@ -1,14 +1,8 @@
 package com.genymobile.scrcpy.wrappers;
 
-import com.genymobile.scrcpy.FakeContext;
 import com.genymobile.scrcpy.Ln;
 
-import android.annotation.SuppressLint;
-import android.annotation.TargetApi;
-import android.content.Intent;
 import android.os.Binder;
-import android.os.Build;
-import android.os.Bundle;
 import android.os.IBinder;
 import android.os.IInterface;
 import android.os.Process;
@@ -17,15 +11,12 @@ import java.lang.reflect.Field;
 import java.lang.reflect.InvocationTargetException;
 import java.lang.reflect.Method;
 
-@SuppressLint("PrivateApi,DiscouragedPrivateApi")
 public class ActivityManager {
 
     private final IInterface manager;
     private Method getContentProviderExternalMethod;
     private boolean getContentProviderExternalMethodNewVersion = true;
     private Method removeContentProviderExternalMethod;
-    private Method startActivityAsUserWithFeatureMethod;
-    private Method forceStopPackageMethod;
 
     public ActivityManager(IInterface manager) {
         this.manager = manager;
@@ -52,7 +43,6 @@ public class ActivityManager {
         return removeContentProviderExternalMethod;
     }
 
-    @TargetApi(Build.VERSION_CODES.Q)
     private ContentProvider getContentProviderExternal(String name, IBinder token) {
         try {
             Method method = getGetContentProviderExternalMethod();
@@ -95,55 +85,4 @@ public class ActivityManager {
     public ContentProvider createSettingsProvider() {
         return getContentProviderExternal("settings", new Binder());
     }
-
-    private Method getStartActivityAsUserWithFeatureMethod() throws NoSuchMethodException, ClassNotFoundException {
-        if (startActivityAsUserWithFeatureMethod == null) {
-            Class<?> iApplicationThreadClass = Class.forName("android.app.IApplicationThread");
-            Class<?> profilerInfo = Class.forName("android.app.ProfilerInfo");
-            startActivityAsUserWithFeatureMethod = manager.getClass()
-                    .getMethod("startActivityAsUserWithFeature", iApplicationThreadClass, String.class, String.class, Intent.class, String.class,
-                            IBinder.class, String.class, int.class, int.class, profilerInfo, Bundle.class, int.class);
-        }
-        return startActivityAsUserWithFeatureMethod;
-    }
-
-    @SuppressWarnings("ConstantConditions")
-    public int startActivityAsUserWithFeature(Intent intent) {
-        try {
-            Method method = getStartActivityAsUserWithFeatureMethod();
-            return (int) method.invoke(
-                    /* this */ manager,
-                    /* caller */ null,
-                    /* callingPackage */ FakeContext.PACKAGE_NAME,
-                    /* callingFeatureId */ null,
-                    /* intent */ intent,
-                    /* resolvedType */ null,
-                    /* resultTo */ null,
-                    /* resultWho */ null,
-                    /* requestCode */ 0,
-                    /* startFlags */ 0,
-                    /* profilerInfo */ null,
-                    /* bOptions */ null,
-                    /* userId */ /* UserHandle.USER_CURRENT */ -2);
-        } catch (Throwable e) {
-            Ln.e("Could not invoke method", e);
-            return 0;
-        }
-    }
-
-    private Method getForceStopPackageMethod() throws NoSuchMethodException {
-        if (forceStopPackageMethod == null) {
-            forceStopPackageMethod = manager.getClass().getMethod("forceStopPackage", String.class, int.class);
-        }
-        return forceStopPackageMethod;
-    }
-
-    public void forceStopPackage(String packageName) {
-        try {
-            Method method = getForceStopPackageMethod();
-            method.invoke(manager, packageName, /* userId */ /* UserHandle.USER_CURRENT */ -2);
-        } catch (Throwable e) {
-            Ln.e("Could not invoke method", e);
-        }
-    }
 }