Reset video capture on folding event

Handle folding event the same way as rotation events.

Fixes #3960 <https://github.com/Genymobile/scrcpy/issues/3960>
PR #3979 <https://github.com/Genymobile/scrcpy/pull/3979>

Signed-off-by: Romain Vimont <rom@rom1v.com>

app/data/bash-completion/scrcpy

@@ -7,7 +7,6 @@ _scrcpy() {
         --audio-codec=
         --audio-codec-options=
         --audio-encoder=
-        --audio-source=
         --audio-output-buffer=
         -b --video-bit-rate=
         --crop=
@@ -87,10 +86,6 @@ _scrcpy() {
             COMPREPLY=($(compgen -W 'opus aac raw' -- "$cur"))
             return
             ;;
-        --audio-source)
-            COMPREPLY=($(compgen -W 'output mic' -- "$cur"))
-            return
-            ;;
         --lock-video-orientation)
             COMPREPLY=($(compgen -W 'unlocked initial 0 1 2 3' -- "$cur"))
             return

app/data/zsh-completion/_scrcpy

@@ -14,7 +14,6 @@ arguments=(
     '--audio-codec=[Select the audio codec]:codec:(opus aac raw)'
     '--audio-codec-options=[Set a list of comma-separated key\:type=value options for the device audio encoder]'
     '--audio-encoder=[Use a specific MediaCodec audio encoder]'
-    '--audio-source=[Select the audio source]:source:(output mic)'
     '--audio-output-buffer=[Configure the size of the SDL audio output buffer (in milliseconds)]'
     {-b,--video-bit-rate=}'[Encode the video at the given bit-rate]'
     '--crop=[\[width\:height\:x\:y\] Crop the device screen on the server]'

app/scrcpy.1

@@ -33,6 +33,14 @@ Lower values decrease the latency, but increase the likelyhood of buffer underru
 
 Default is 50.
 
+.TP
+.BI "\-\-audio\-output\-buffer ms
+Configure the size of the SDL audio output buffer (in milliseconds).
+
+If you get "robotic" audio playback, you should test with a higher value (10). Do not change this setting otherwise.
+
+Default is 5.
+
 .TP
 .BI "\-\-audio\-codec " name
 Select an audio codec (opus, aac or raw).
@@ -55,20 +63,6 @@ Use a specific MediaCodec audio encoder (depending on the codec provided by \fB\
 
 The available encoders can be listed by \-\-list\-encoders.
 
-.TP
-.BI "\-\-audio\-source " source
-Select the audio source (output or mic).
-
-Default is output.
-
-.TP
-.BI "\-\-audio\-output\-buffer ms
-Configure the size of the SDL audio output buffer (in milliseconds).
-
-If you get "robotic" audio playback, you should test with a higher value (10). Do not change this setting otherwise.
-
-Default is 5.
-
 .TP
 .BI "\-b, \-\-video\-bit\-rate " value
 Encode the video at the given bit\-rate, expressed in bits/s. Unit suffixes are supported: '\fBK\fR' (x1000) and '\fBM\fR' (x1000000).

app/src/audio_player.c

@@ -107,7 +107,7 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
         // latency.
         LOGD("[Audio] Buffer underflow, inserting silence: %" PRIu32 " samples",
              silence);
-        memset(stream + TO_BYTES(read), 0, TO_BYTES(silence));
+        memset(stream + read, 0, TO_BYTES(silence));
 
         if (ap->received) {
             // Inserting additional samples immediately increases buffering

app/src/cli.c

@@ -76,7 +76,6 @@ enum {
     OPT_NO_VIDEO,
     OPT_NO_AUDIO_PLAYBACK,
     OPT_NO_VIDEO_PLAYBACK,
-    OPT_AUDIO_SOURCE,
 };
 
 struct sc_option {
@@ -135,6 +134,16 @@ static const struct sc_option options[] = {
                 "likelyhood of buffer underrun (causing audio glitches).\n"
                 "Default is 50.",
     },
+    {
+        .longopt_id = OPT_AUDIO_OUTPUT_BUFFER,
+        .longopt = "audio-output-buffer",
+        .argdesc = "ms",
+        .text = "Configure the size of the SDL audio output buffer (in "
+                "milliseconds).\n"
+                "If you get \"robotic\" audio playback, you should test with "
+                "a higher value (10). Do not change this setting otherwise.\n"
+                "Default is 5.",
+    },
     {
         .longopt_id = OPT_AUDIO_CODEC,
         .longopt = "audio-codec",
@@ -162,23 +171,6 @@ static const struct sc_option options[] = {
                 "codec provided by --audio-codec).\n"
                 "The available encoders can be listed by --list-encoders.",
     },
-    {
-        .longopt_id = OPT_AUDIO_SOURCE,
-        .longopt = "audio-source",
-        .argdesc = "source",
-        .text = "Select the audio source (output or mic).\n"
-                "Default is output.",
-    },
-    {
-        .longopt_id = OPT_AUDIO_OUTPUT_BUFFER,
-        .longopt = "audio-output-buffer",
-        .argdesc = "ms",
-        .text = "Configure the size of the SDL audio output buffer (in "
-                "milliseconds).\n"
-                "If you get \"robotic\" audio playback, you should test with "
-                "a higher value (10). Do not change this setting otherwise.\n"
-                "Default is 5.",
-    },
     {
         .shortopt = 'b',
         .longopt = "video-bit-rate",
@@ -1596,22 +1588,6 @@ parse_audio_codec(const char *optarg, enum sc_codec *codec) {
     return false;
 }
 
-static bool
-parse_audio_source(const char *optarg, enum sc_audio_source *source) {
-    if (!strcmp(optarg, "mic")) {
-        *source = SC_AUDIO_SOURCE_MIC;
-        return true;
-    }
-
-    if (!strcmp(optarg, "output")) {
-        *source = SC_AUDIO_SOURCE_OUTPUT;
-        return true;
-    }
-
-    LOGE("Unsupported audio source: %s (expected output or mic)", optarg);
-    return false;
-}
-
 static bool
 parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
                        const char *optstring, const struct option *longopts) {
@@ -1939,11 +1915,6 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
                     return false;
                 }
                 break;
-            case OPT_AUDIO_SOURCE:
-                if (!parse_audio_source(optarg, &opts->audio_source)) {
-                    return false;
-                }
-                break;
             default:
                 // getopt prints the error message on stderr
                 return false;

app/src/demuxer.c

@@ -79,8 +79,9 @@ sc_demuxer_recv_video_size(struct sc_demuxer *demuxer, uint32_t *width,
 
 static bool
 sc_demuxer_recv_packet(struct sc_demuxer *demuxer, AVPacket *packet) {
-    // The video and audio streams contain a sequence of raw packets (as
-    // provided by MediaCodec), each prefixed with a "meta" header.
+    // The video stream contains raw packets, without time information. When we
+    // record, we retrieve the timestamps separately, from a "meta" header
+    // added by the server before each raw packet.
     //
     // The "meta" header length is 12 bytes:
     // [. . . . . . . .|. . . .]. . . . . . . . . . . . . . . ...

app/src/options.c

@@ -14,7 +14,6 @@ const struct scrcpy_options scrcpy_options_default = {
     .log_level = SC_LOG_LEVEL_INFO,
     .video_codec = SC_CODEC_H264,
     .audio_codec = SC_CODEC_OPUS,
-    .audio_source = SC_AUDIO_SOURCE_OUTPUT,
     .record_format = SC_RECORD_FORMAT_AUTO,
     .keyboard_input_mode = SC_KEYBOARD_INPUT_MODE_INJECT,
     .mouse_input_mode = SC_MOUSE_INPUT_MODE_INJECT,

app/src/options.h

@@ -44,11 +44,6 @@ enum sc_codec {
     SC_CODEC_RAW,
 };
 
-enum sc_audio_source {
-    SC_AUDIO_SOURCE_OUTPUT,
-    SC_AUDIO_SOURCE_MIC,
-};
-
 enum sc_lock_video_orientation {
     SC_LOCK_VIDEO_ORIENTATION_UNLOCKED = -1,
     // lock the current orientation when scrcpy starts
@@ -120,7 +115,6 @@ struct scrcpy_options {
     enum sc_log_level log_level;
     enum sc_codec video_codec;
     enum sc_codec audio_codec;
-    enum sc_audio_source audio_source;
     enum sc_record_format record_format;
     enum sc_keyboard_input_mode keyboard_input_mode;
     enum sc_mouse_input_mode mouse_input_mode;

app/src/scrcpy.c

@@ -334,7 +334,6 @@ scrcpy(struct scrcpy_options *options) {
         .log_level = options->log_level,
         .video_codec = options->video_codec,
         .audio_codec = options->audio_codec,
-        .audio_source = options->audio_source,
         .crop = options->crop,
         .port_range = options->port_range,
         .tunnel_host = options->tunnel_host,

app/src/server.c

@@ -246,10 +246,6 @@ execute_server(struct sc_server *server,
         ADD_PARAM("audio_codec=%s",
             sc_server_get_codec_name(params->audio_codec));
     }
-    if (params->audio_source != SC_AUDIO_SOURCE_OUTPUT) {
-        assert(params->audio_source == SC_AUDIO_SOURCE_MIC);
-        ADD_PARAM("audio_source=mic");
-    }
     if (params->max_size) {
         ADD_PARAM("max_size=%" PRIu16, params->max_size);
     }

app/src/server.h

@@ -26,7 +26,6 @@ struct sc_server_params {
     enum sc_log_level log_level;
     enum sc_codec video_codec;
     enum sc_codec audio_codec;
-    enum sc_audio_source audio_source;
     const char *crop;
     const char *video_codec_options;
     const char *audio_codec_options;

doc/audio.md

@@ -41,24 +41,6 @@ interesting to add [buffering](#buffering) to minimize glitches:
 scrcpy --no-video --audio-buffer=200
 ```
 
-## Source
-
-By default, the device audio output is forwarded.
-
-It is possible to capture the device microphone instead:
-
-```
-scrcpy --audio-source=mic
-```
-
-For example, to use the device as a dictaphone and record a capture directly on
-the computer:
-
-```
-scrcpy --audio-source=mic --no-video --no-audio-playback --record=file.opus
-```
-
-
 ## Codec
 
 The audio codec can be selected. The possible values are `opus` (default), `aac`

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

@@ -10,6 +10,7 @@ import android.media.AudioFormat;
 import android.media.AudioRecord;
 import android.media.AudioTimestamp;
 import android.media.MediaCodec;
+import android.media.MediaRecorder;
 import android.os.Build;
 import android.os.SystemClock;
 
@@ -17,6 +18,7 @@ import java.nio.ByteBuffer;
 
 public final class AudioCapture {
 
+    public static final int SOURCE = MediaRecorder.AudioSource.REMOTE_SUBMIX;
     public static final int SAMPLE_RATE = 48000;
     public static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_STEREO;
     public static final int CHANNELS = 2;
@@ -24,18 +26,12 @@ public final class AudioCapture {
     public static final int ENCODING = AudioFormat.ENCODING_PCM_16BIT;
     public static final int BYTES_PER_SAMPLE = 2;
 
-    private final int audioSource;
-
     private AudioRecord recorder;
 
     private final AudioTimestamp timestamp = new AudioTimestamp();
     private long previousPts = 0;
     private long nextPts = 0;
 
-    public AudioCapture(AudioSource audioSource) {
-        this.audioSource = audioSource.value();
-    }
-
     public static int millisToBytes(int millis) {
         return SAMPLE_RATE * CHANNELS * BYTES_PER_SAMPLE * millis / 1000;
     }
@@ -50,13 +46,13 @@ public final class AudioCapture {
 
     @TargetApi(Build.VERSION_CODES.M)
     @SuppressLint({"WrongConstant", "MissingPermission"})
-    private static AudioRecord createAudioRecord(int audioSource) {
+    private static AudioRecord createAudioRecord() {
         AudioRecord.Builder builder = new AudioRecord.Builder();
         if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.S) {
             // On older APIs, Workarounds.fillAppInfo() must be called beforehand
             builder.setContext(FakeContext.get());
         }
-        builder.setAudioSource(audioSource);
+        builder.setAudioSource(SOURCE);
         builder.setAudioFormat(createAudioFormat());
         int minBufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_CONFIG, ENCODING);
         // This buffer size does not impact latency
@@ -104,12 +100,12 @@ public final class AudioCapture {
 
     private void startRecording() {
         try {
-            recorder = createAudioRecord(audioSource);
+            recorder = createAudioRecord();
         } catch (NullPointerException e) {
             // Creating an AudioRecord using an AudioRecord.Builder does not work on Vivo phones:
             // - <https://github.com/Genymobile/scrcpy/issues/3805>
             // - <https://github.com/Genymobile/scrcpy/pull/3862>
-            recorder = Workarounds.createAudioRecord(audioSource, SAMPLE_RATE, CHANNEL_CONFIG, CHANNELS, CHANNEL_MASK, ENCODING);
+            recorder = Workarounds.createAudioRecord(SOURCE, SAMPLE_RATE, CHANNEL_CONFIG, CHANNELS, CHANNEL_MASK, ENCODING);
         }
         recorder.startRecording();
     }

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

@@ -40,7 +40,6 @@ public final class AudioEncoder implements AsyncProcessor {
     private static final int READ_MS = 5; // milliseconds
     private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
 
-    private final AudioCapture capture;
     private final Streamer streamer;
     private final int bitRate;
     private final List<CodecOption> codecOptions;
@@ -59,8 +58,7 @@ public final class AudioEncoder implements AsyncProcessor {
 
     private boolean ended;
 
-    public AudioEncoder(AudioCapture capture, Streamer streamer, int bitRate, List<CodecOption> codecOptions, String encoderName) {
-        this.capture = capture;
+    public AudioEncoder(Streamer streamer, int bitRate, List<CodecOption> codecOptions, String encoderName) {
         this.streamer = streamer;
         this.bitRate = bitRate;
         this.codecOptions = codecOptions;
@@ -177,6 +175,7 @@ public final class AudioEncoder implements AsyncProcessor {
         }
 
         MediaCodec mediaCodec = null;
+        AudioCapture capture = new AudioCapture();
 
         boolean mediaCodecStarted = false;
         try {
@@ -193,9 +192,10 @@ public final class AudioEncoder implements AsyncProcessor {
             capture.start();
 
             final MediaCodec mediaCodecRef = mediaCodec;
+            final AudioCapture captureRef = capture;
             inputThread = new Thread(() -> {
                 try {
-                    inputThread(mediaCodecRef, capture);
+                    inputThread(mediaCodecRef, captureRef);
                 } catch (IOException | InterruptedException e) {
                     Ln.e("Audio capture error", e);
                 } finally {

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

@@ -8,7 +8,6 @@ import java.nio.ByteBuffer;
 
 public final class AudioRawRecorder implements AsyncProcessor {
 
-    private final AudioCapture capture;
     private final Streamer streamer;
 
     private Thread thread;
@@ -16,8 +15,7 @@ public final class AudioRawRecorder implements AsyncProcessor {
     private static final int READ_MS = 5; // milliseconds
     private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
 
-    public AudioRawRecorder(AudioCapture capture, Streamer streamer) {
-        this.capture = capture;
+    public AudioRawRecorder(Streamer streamer) {
         this.streamer = streamer;
     }
 
@@ -31,6 +29,7 @@ public final class AudioRawRecorder implements AsyncProcessor {
         final ByteBuffer buffer = ByteBuffer.allocateDirect(READ_SIZE);
         final MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
 
+        AudioCapture capture = new AudioCapture();
         try {
             capture.start();
 

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

@@ -1,30 +0,0 @@
-package com.genymobile.scrcpy;
-
-import android.media.MediaRecorder;
-
-public enum AudioSource {
-    OUTPUT("output", MediaRecorder.AudioSource.REMOTE_SUBMIX),
-    MIC("mic", MediaRecorder.AudioSource.MIC);
-
-    private final String name;
-    private final int value;
-
-    AudioSource(String name, int value) {
-        this.name = name;
-        this.value = value;
-    }
-
-    int value() {
-        return value;
-    }
-
-    static AudioSource findByName(String name) {
-        for (AudioSource audioSource : AudioSource.values()) {
-            if (name.equals(audioSource.name)) {
-                return audioSource;
-            }
-        }
-
-        return null;
-    }
-}

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

@@ -14,7 +14,6 @@ public class Options {
     private int maxSize;
     private VideoCodec videoCodec = VideoCodec.H264;
     private AudioCodec audioCodec = AudioCodec.OPUS;
-    private AudioSource audioSource = AudioSource.OUTPUT;
     private int videoBitRate = 8000000;
     private int audioBitRate = 128000;
     private int maxFps;
@@ -73,10 +72,6 @@ public class Options {
         return audioCodec;
     }
 
-    public AudioSource getAudioSource() {
-        return audioSource;
-    }
-
     public int getVideoBitRate() {
         return videoBitRate;
     }
@@ -230,13 +225,6 @@ public class Options {
                     }
                     options.audioCodec = audioCodec;
                     break;
-                case "audio_source":
-                    AudioSource audioSource = AudioSource.findByName(value);
-                    if (audioSource == null) {
-                        throw new IllegalArgumentException("Audio source " + value + " not supported");
-                    }
-                    options.audioSource = audioSource;
-                    break;
                 case "max_size":
                     options.maxSize = Integer.parseInt(value) & ~7; // multiple of 8
                     break;

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

@@ -136,14 +136,13 @@ public final class Server {
 
             if (audio) {
                 AudioCodec audioCodec = options.getAudioCodec();
-                AudioCapture audioCapture = new AudioCapture(options.getAudioSource());
                 Streamer audioStreamer = new Streamer(connection.getAudioFd(), audioCodec, options.getSendCodecMeta(),
                         options.getSendFrameMeta());
                 AsyncProcessor audioRecorder;
                 if (audioCodec == AudioCodec.RAW) {
-                    audioRecorder = new AudioRawRecorder(audioCapture, audioStreamer);
+                    audioRecorder = new AudioRawRecorder(audioStreamer);
                 } else {
-                    audioRecorder = new AudioEncoder(audioCapture, audioStreamer, options.getAudioBitRate(), options.getAudioCodecOptions(),
+                    audioRecorder = new AudioEncoder(audioStreamer, options.getAudioBitRate(), options.getAudioCodecOptions(),
                             options.getAudioEncoder());
                 }
                 asyncProcessors.add(audioRecorder);