Add compat support for FFmpeg < 5.1

The new chlayout API has been introduced in FFmpeg 5.1. Use the old
channel_layout API on older versions.

app/meson.build

@@ -4,6 +4,7 @@ src = [
     'src/adb/adb_device.c',
     'src/adb/adb_parser.c',
     'src/adb/adb_tunnel.c',
+    'src/audio_player.c',
     'src/cli.c',
     'src/clock.c',
     'src/compat.c',
@@ -30,6 +31,7 @@ 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',
@@ -100,6 +102,7 @@ if not crossbuild_windows
         dependency('libavformat', version: '>= 57.33'),
         dependency('libavcodec', version: '>= 57.37'),
         dependency('libavutil'),
+        dependency('libswresample'),
         dependency('sdl2', version: '>= 2.0.5'),
     ]
 
@@ -134,12 +137,14 @@ else
     ffmpeg_avcodec = meson.get_cross_property('ffmpeg_avcodec')
     ffmpeg_avformat = meson.get_cross_property('ffmpeg_avformat')
     ffmpeg_avutil = meson.get_cross_property('ffmpeg_avutil')
+    ffmpeg_swresample = meson.get_cross_property('ffmpeg_swresample')
 
     ffmpeg = declare_dependency(
         dependencies: [
             cc.find_library(ffmpeg_avcodec, dirs: ffmpeg_bin_dir),
             cc.find_library(ffmpeg_avformat, dirs: ffmpeg_bin_dir),
             cc.find_library(ffmpeg_avutil, dirs: ffmpeg_bin_dir),
+            cc.find_library(ffmpeg_swresample, dirs: ffmpeg_bin_dir),
         ],
         include_directories: include_directories(ffmpeg_include_dir)
     )

app/src/audio_player.c

@@ -0,0 +1,304 @@
+#include "audio_player.h"
+
+#include <libavutil/opt.h>
+
+#include "util/log.h"
+
+#define SC_AUDIO_PLAYER_NDEBUG // comment to debug
+
+/** Downcast frame_sink to sc_audio_player */
+#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
+
+void
+sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
+    struct sc_audio_player *ap = userdata;
+
+    // This callback is called with the lock used by SDL_AudioDeviceLock(), so
+    // the bytebuf is protected
+
+    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) {
+    assert(ap->nb_channels);
+    assert(ap->out_bytes_per_sample);
+    return samples * ap->nb_channels * ap->out_bytes_per_sample;
+}
+
+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);
+    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);
+        if (!buf) {
+            LOG_OOM();
+            // Could not realloc to the requested size
+            return NULL;
+        }
+        ap->swr_buf = buf;
+        ap->swr_buf_alloc_size = new_size;
+    }
+
+    return ap->swr_buf;
+}
+
+static bool
+sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
+                                const AVCodecContext *ctx) {
+    struct sc_audio_player *ap = DOWNCAST(sink);
+#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
+    assert(ctx->ch_layout.nb_channels > 0);
+    unsigned nb_channels = ctx->ch_layout.nb_channels;
+#else
+    int tmp = av_get_channel_layout_nb_channels(ctx->channel_layout);
+    assert(tmp > 0);
+    unsigned nb_channels = tmp;
+#endif
+
+    SDL_AudioSpec desired = {
+        .freq = ctx->sample_rate,
+        .format = SC_SDL_SAMPLE_FMT,
+        .channels = 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;
+
+    assert(ctx->sample_rate > 0);
+    assert(!av_sample_fmt_is_planar(SC_AV_SAMPLE_FMT));
+
+    int out_bytes_per_sample = av_get_bytes_per_sample(SC_AV_SAMPLE_FMT);
+    assert(out_bytes_per_sample > 0);
+
+#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
+    av_opt_set_chlayout(swr_ctx, "in_chlayout", &ctx->ch_layout, 0);
+    av_opt_set_chlayout(swr_ctx, "out_chlayout", &ctx->ch_layout, 0);
+#else
+    av_opt_set_channel_layout(swr_ctx, "in_channel_layout",
+                              ctx->channel_layout, 0);
+    av_opt_set_channel_layout(swr_ctx, "out_channel_layout",
+                              ctx->channel_layout, 0);
+#endif
+
+    av_opt_set_int(swr_ctx, "in_sample_rate", ctx->sample_rate, 0);
+    av_opt_set_int(swr_ctx, "out_sample_rate", ctx->sample_rate, 0);
+
+    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", ctx->sample_fmt, 0);
+    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", SC_AV_SAMPLE_FMT, 0);
+
+    int ret = swr_init(swr_ctx);
+    if (ret) {
+        LOGE("Failed to initialize the resampling context");
+        goto error_free_swr_ctx;
+    }
+
+    ap->sample_rate = ctx->sample_rate;
+    ap->nb_channels = 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);
+    ap->swr_buf = malloc(initial_swr_buf_size);
+    if (!ap->swr_buf) {
+        LOG_OOM();
+        goto error_destroy_bytebuf;
+    }
+    ap->swr_buf_alloc_size = initial_swr_buf_size;
+
+    sc_average_init(&ap->avg_buffered_samples, 32);
+    ap->samples_since_resync = 0;
+
+    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
+sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
+    struct sc_audio_player *ap = DOWNCAST(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
+sc_audio_player_frame_sink_push(struct sc_frame_sink *sink, const AVFrame *frame) {
+    struct sc_audio_player *ap = DOWNCAST(sink);
+
+    SwrContext *swr_ctx = ap->swr_ctx;
+
+    int64_t delay = swr_get_delay(swr_ctx, ap->sample_rate);
+    // No need to av_rescale_rnd(), input and output sample rates are the same
+    int dst_nb_samples = delay + frame->nb_samples;
+
+    uint8_t *swr_buf = sc_audio_player_get_swr_buf(ap, frame->nb_samples);
+    if (!swr_buf) {
+        return false;
+    }
+
+    int ret = swr_convert(swr_ctx, &swr_buf, dst_nb_samples,
+                          (const uint8_t **) frame->data, frame->nb_samples);
+    if (ret < 0) {
+        LOGE("Resampling failed: %d", ret);
+        return false;
+    }
+
+    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
+
+    // It should almost always be possible to write without lock
+    bool can_write_without_lock = swr_buf_size <= ap->safe_empty_buffer;
+    if (can_write_without_lock) {
+        sc_bytebuf_prepare_write(&ap->buf, swr_buf, swr_buf_size);
+    }
+
+    SDL_LockAudioDevice(ap->device);
+    if (can_write_without_lock) {
+        sc_bytebuf_commit_write(&ap->buf, swr_buf_size);
+    } else {
+        sc_bytebuf_write(&ap->buf, swr_buf, swr_buf_size);
+    }
+
+    // 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) {
+    static const struct sc_frame_sink_ops ops = {
+        .open = sc_audio_player_frame_sink_open,
+        .close = sc_audio_player_frame_sink_close,
+        .push = sc_audio_player_frame_sink_push,
+    };
+
+    ap->frame_sink.ops = &ops;
+}

app/src/audio_player.h

@@ -0,0 +1,54 @@
+#ifndef SC_AUDIO_PLAYER_H
+#define SC_AUDIO_PLAYER_H
+
+#include "common.h"
+
+#include <stdbool.h>
+#include "trait/frame_sink.h"
+#include <util/average.h>
+#include <util/bytebuf.h>
+#include <util/thread.h>
+
+#include <libavformat/avformat.h>
+#include <libswresample/swresample.h>
+#include <SDL2/SDL.h>
+
+struct sc_audio_player {
+    struct sc_frame_sink frame_sink;
+
+    SDL_AudioDeviceID device;
+
+    // protected by SDL_AudioDeviceLock()
+    struct sc_bytebuf buf;
+    // Number of bytes which could be written without locking
+    size_t safe_empty_buffer;
+
+    struct SwrContext *swr_ctx;
+
+    // The sample rate is the same for input and output
+    unsigned sample_rate;
+    // The number of channels is the same for input and output
+    unsigned nb_channels;
+
+    unsigned out_bytes_per_sample;
+
+    // Target buffer for resampling
+    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;
+};
+
+struct sc_audio_player_callbacks {
+    void (*on_ended)(struct sc_audio_player *ap, bool success, void *userdata);
+};
+
+void
+sc_audio_player_init(struct sc_audio_player *ap);
+
+#endif

app/src/compat.h

@@ -37,6 +37,13 @@
 # define SCRCPY_LAVF_HAS_AVFORMATCONTEXT_URL
 #endif
 
+// Not documented in ffmpeg/doc/APIchanges, but the channel_layout API
+// has been replaced by chlayout in FFmpeg commit
+// f423497b455da06c1337846902c770028760e094.
+#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 23, 100)
+# define SCRCPY_LAVU_HAS_CHLAYOUT
+#endif
+
 #if SDL_VERSION_ATLEAST(2, 0, 6)
 // <https://github.com/libsdl-org/SDL/commit/d7a318de563125e5bb465b1000d6bc9576fbc6fc>
 # define SCRCPY_SDL_HAS_HINT_TOUCH_MOUSE_EVENTS

app/src/decoder.c

@@ -2,6 +2,7 @@
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavutil/channel_layout.h>
 
 #include "events.h"
 #include "video_buffer.h"
@@ -50,6 +51,16 @@ sc_decoder_open(struct sc_decoder *decoder, const AVCodec *codec) {
     if (codec->type == AVMEDIA_TYPE_VIDEO) {
         // Hardcoded video properties
         decoder->codec_ctx->pix_fmt = AV_PIX_FMT_YUV420P;
+    } else {
+        // Hardcoded audio properties
+#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
+        decoder->codec_ctx->ch_layout =
+            (AVChannelLayout) AV_CHANNEL_LAYOUT_STEREO;
+#else
+        decoder->codec_ctx->channel_layout = AV_CH_LAYOUT_STEREO;
+        decoder->codec_ctx->channels = 2;
+#endif
+        decoder->codec_ctx->sample_rate = 48000;
     }
 
     if (avcodec_open2(decoder->codec_ctx, codec, NULL) < 0) {

app/src/recorder.c

@@ -216,7 +216,12 @@ sc_recorder_wait_audio_stream(struct sc_recorder *recorder) {
 
         stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
         stream->codecpar->codec_id = codec->id;
+#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
         stream->codecpar->ch_layout.nb_channels = 2;
+#else
+        stream->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
+        stream->codecpar->channels = 2;
+#endif
         stream->codecpar->sample_rate = 48000;
 
         recorder->audio_stream_index = stream->index;

app/src/scrcpy.c

@@ -13,6 +13,7 @@
 # include <windows.h>
 #endif
 
+#include "audio_player.h"
 #include "controller.h"
 #include "decoder.h"
 #include "demuxer.h"
@@ -40,6 +41,7 @@
 struct scrcpy {
     struct sc_server server;
     struct sc_screen screen;
+    struct sc_audio_player audio_player;
     struct sc_demuxer video_demuxer;
     struct sc_demuxer audio_demuxer;
     struct sc_decoder video_decoder;
@@ -383,9 +385,16 @@ scrcpy(struct scrcpy_options *options) {
     }
 
     // Initialize SDL video in addition if display is enabled
-    if (options->display && SDL_Init(SDL_INIT_VIDEO)) {
-        LOGE("Could not initialize SDL: %s", SDL_GetError());
-        goto end;
+    if (options->display) {
+        if (SDL_Init(SDL_INIT_VIDEO)) {
+            LOGE("Could not initialize SDL video: %s", SDL_GetError());
+            goto end;
+        }
+
+        if (options->audio && SDL_Init(SDL_INIT_AUDIO)) {
+            LOGE("Could not initialize SDL audio: %s", SDL_GetError());
+            goto end;
+        }
     }
 
     sdl_configure(options->display, options->disable_screensaver);
@@ -663,6 +672,11 @@ aoa_hid_end:
         screen_initialized = true;
 
         sc_decoder_add_sink(&s->video_decoder, &s->screen.frame_sink);
+
+        if (options->audio) {
+            sc_audio_player_init(&s->audio_player);
+            sc_decoder_add_sink(&s->audio_decoder, &s->audio_player.frame_sink);
+        }
     }
 
 #ifdef HAVE_V4L2

cross_win32.txt

@@ -19,6 +19,7 @@ endian = 'little'
 ffmpeg_avcodec = 'avcodec-58'
 ffmpeg_avformat = 'avformat-58'
 ffmpeg_avutil = 'avutil-56'
+ffmpeg_swresample = 'swresample-3'
 prebuilt_ffmpeg = 'ffmpeg-win32-4.3.1'
 prebuilt_sdl2 = 'SDL2-2.26.1/i686-w64-mingw32'
 prebuilt_libusb_root = 'libusb-1.0.26'

cross_win64.txt

@@ -19,6 +19,7 @@ endian = 'little'
 ffmpeg_avcodec = 'avcodec-59'
 ffmpeg_avformat = 'avformat-59'
 ffmpeg_avutil = 'avutil-57'
+ffmpeg_swresample = 'swresample-4'
 prebuilt_ffmpeg = 'ffmpeg-win64-5.1.2'
 prebuilt_sdl2 = 'SDL2-2.26.1/x86_64-w64-mingw32'
 prebuilt_libusb_root = 'libusb-1.0.26'

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

@@ -1,7 +1,11 @@
 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;
@@ -12,6 +16,7 @@ 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;
@@ -215,6 +220,32 @@ 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) {
@@ -240,8 +271,19 @@ public final class AudioEncoder {
             mediaCodec.setCallback(new EncoderCallback(), new Handler(mediaCodecThread.getLooper()));
             mediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
 
-            recorder = createAudioRecord();
-            recorder.startRecording();
+            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();
+            }
             recorderStarted = true;
 
             final MediaCodec mediaCodecRef = mediaCodec;

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

@@ -3,7 +3,12 @@ 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;
 
@@ -11,12 +16,15 @@ 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;
@@ -43,6 +51,7 @@ public class ActivityManager {
         return removeContentProviderExternalMethod;
     }
 
+    @TargetApi(Build.VERSION_CODES.Q)
     private ContentProvider getContentProviderExternal(String name, IBinder token) {
         try {
             Method method = getGetContentProviderExternalMethod();
@@ -85,4 +94,55 @@ 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);
+        }
+    }
 }