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3 Commits
Author | SHA1 | Date | |
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14348f42b2 | |||
b5ed834012 | |||
65b9f04f39 |
@ -66,6 +66,8 @@ static void SDLCALL
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sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
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struct sc_audio_player *ap = userdata;
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// This callback is called with the lock used by SDL_LockAudioDevice()
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assert(len_int > 0);
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size_t len = len_int;
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uint32_t count = TO_SAMPLES(len);
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@ -179,19 +181,29 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
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if (written < samples) {
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uint32_t remaining = samples - written;
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assert(remaining <= cap);
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skipped_samples = sc_audiobuf_truncate(&ap->buf, cap - remaining);
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// All samples that could be written without locking have been written,
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// now we need to lock to drop/consume old samples
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SDL_LockAudioDevice(ap->device);
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LOGW("Audio buffer full, %" PRIu32 " samples dropped", skipped_samples);
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// Retry with the lock
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written += sc_audiobuf_write(&ap->buf,
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swr_buf + TO_BYTES(written),
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remaining);
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if (written < samples) {
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remaining = samples - written;
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// Still insufficient, drop old samples to make space
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skipped_samples = sc_audiobuf_read(&ap->buf, NULL, remaining);
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assert(skipped_samples == remaining);
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// Now there is enough space
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uint32_t w = sc_audiobuf_write(&ap->buf,
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swr_buf + TO_BYTES(written),
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remaining);
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assert(w == remaining);
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(void) w;
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// Now there is enough space
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uint32_t w = sc_audiobuf_write(&ap->buf,
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swr_buf + TO_BYTES(written),
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remaining);
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assert(w == remaining);
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(void) w;
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}
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written = samples;
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SDL_UnlockAudioDevice(ap->device);
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}
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uint32_t underflow = 0;
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@ -213,22 +225,30 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
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uint32_t can_read = sc_audiobuf_can_read(&ap->buf);
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if (can_read > max_buffered_samples) {
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uint32_t skipped = sc_audiobuf_truncate(&ap->buf, max_buffered_samples);
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assert(skipped);
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if (played) {
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LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
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" samples", skipped);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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} else {
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LOGD("[Audio] Playback not started, skipping %" PRIu32 " samples",
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skipped);
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#endif
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}
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skipped_samples += skipped;
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uint32_t skip_samples = 0;
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SDL_LockAudioDevice(ap->device);
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can_read = sc_audiobuf_can_read(&ap->buf);
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if (can_read > max_buffered_samples) {
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skip_samples = can_read - max_buffered_samples;
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uint32_t r = sc_audiobuf_read(&ap->buf, NULL, skip_samples);
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assert(r == skip_samples);
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(void) r;
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skipped_samples += skip_samples;
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}
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SDL_UnlockAudioDevice(ap->device);
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if (skip_samples) {
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if (played) {
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LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
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" samples", skip_samples);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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} else {
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LOGD("[Audio] Playback not started, skipping %" PRIu32
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" samples", skip_samples);
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#endif
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}
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}
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}
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atomic_store_explicit(&ap->received, true, memory_order_relaxed);
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@ -388,7 +408,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
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// Use a ring-buffer of the target buffering size plus 1 second between the
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// producer and the consumer. It's too big on purpose, to guarantee that
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// the producer and the consumer will be able to access it in parallel
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// without dropping samples.
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// without locking.
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uint32_t audiobuf_samples = ap->target_buffering + ap->sample_rate;
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size_t sample_size = ap->nb_channels * ap->out_bytes_per_sample;
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@ -2738,11 +2738,6 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
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}
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if (opts->record_filename) {
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if (!opts->video && !opts->audio) {
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LOGE("Video and audio disabled, nothing to record");
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return false;
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}
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if (!opts->record_format) {
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opts->record_format = guess_record_format(opts->record_filename);
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if (!opts->record_format) {
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@ -37,10 +37,10 @@ sc_audiobuf_read(struct sc_audiobuf *buf, void *to_, uint32_t samples_count) {
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assert(samples_count);
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uint8_t *to = to_;
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assert(to);
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// The tail cursor may be updated by the writer thread to drop samples
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uint32_t tail = atomic_load_explicit(&buf->tail, memory_order_acquire);
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// Only the reader thread can write tail without synchronization, so
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// memory_order_relaxed is sufficient
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uint32_t tail = atomic_load_explicit(&buf->tail, memory_order_relaxed);
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// The head cursor is updated after the data is written to the array
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uint32_t head = atomic_load_explicit(&buf->head, memory_order_acquire);
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@ -50,19 +50,21 @@ sc_audiobuf_read(struct sc_audiobuf *buf, void *to_, uint32_t samples_count) {
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samples_count = can_read;
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}
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uint32_t right_count = buf->alloc_size - tail;
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if (right_count > samples_count) {
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right_count = samples_count;
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}
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memcpy(to,
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buf->data + (tail * buf->sample_size),
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right_count * buf->sample_size);
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if (to) {
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uint32_t right_count = buf->alloc_size - tail;
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if (right_count > samples_count) {
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right_count = samples_count;
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}
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memcpy(to,
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buf->data + (tail * buf->sample_size),
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right_count * buf->sample_size);
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if (samples_count > right_count) {
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uint32_t left_count = samples_count - right_count;
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memcpy(to + (right_count * buf->sample_size),
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buf->data,
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left_count * buf->sample_size);
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if (samples_count > right_count) {
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uint32_t left_count = samples_count - right_count;
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memcpy(to + (right_count * buf->sample_size),
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buf->data,
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left_count * buf->sample_size);
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}
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}
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uint32_t new_tail = (tail + samples_count) % buf->alloc_size;
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@ -108,27 +110,3 @@ sc_audiobuf_write(struct sc_audiobuf *buf, const void *from_,
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return samples_count;
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}
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uint32_t
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sc_audiobuf_truncate(struct sc_audiobuf *buf, uint32_t samples_limit) {
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for (;;) {
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// Only the writer thread can write head, so memory_order_relaxed is
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// sufficient
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uint32_t head = atomic_load_explicit(&buf->head, memory_order_relaxed);
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// The tail cursor is updated after the data is consumed by the reader
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uint32_t tail = atomic_load_explicit(&buf->tail, memory_order_acquire);
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uint32_t can_read = (buf->alloc_size + head - tail) % buf->alloc_size;
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if (can_read <= samples_limit) {
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// Nothing to truncate
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return 0;
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}
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uint32_t skip = can_read - samples_limit;
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uint32_t new_tail = (tail + skip) % buf->alloc_size;
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if (atomic_compare_exchange_weak(&buf->tail, &tail, new_tail)) {
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return skip;
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}
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}
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}
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@ -49,16 +49,6 @@ uint32_t
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sc_audiobuf_write(struct sc_audiobuf *buf, const void *from,
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uint32_t samples_count);
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/**
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* Drop old samples to keep at most sample_limit samples
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*
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* Must be called by the writer thread.
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*
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* \return the number of samples dropped
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*/
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uint32_t
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sc_audiobuf_truncate(struct sc_audiobuf *buf, uint32_t samples_limit);
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static inline uint32_t
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sc_audiobuf_capacity(struct sc_audiobuf *buf) {
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assert(buf->alloc_size);
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@ -33,12 +33,6 @@ scrcpy --no-video --no-audio --keyboard=uhid
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scrcpy --no-video --no-audio -K # short version
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```
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To use AOA instead (over USB only):
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```bash
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scrcpy --no-video --no-audio --keyboard=aoa --mouse=aoa
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```
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## Copy-paste
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@ -39,7 +39,7 @@ It only works if the device is connected over USB.
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See [FAQ](/FAQ.md#otg-issues-on-windows).
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## Control only
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## Control-only
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Note that the purpose of OTG is to control the device without USB debugging
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(adb).
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@ -48,7 +48,7 @@ public class SurfaceEncoder implements AsyncProcessor {
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this.downsizeOnError = downsizeOnError;
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}
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private void streamCapture() throws IOException, ConfigurationException {
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private void streamScreen() throws IOException, ConfigurationException {
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Codec codec = streamer.getCodec();
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MediaCodec mediaCodec = createMediaCodec(codec, encoderName);
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MediaFormat format = createFormat(codec.getMimeType(), videoBitRate, maxFps, codecOptions);
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@ -254,7 +254,7 @@ public class SurfaceEncoder implements AsyncProcessor {
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Looper.prepare();
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try {
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streamCapture();
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streamScreen();
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} catch (ConfigurationException e) {
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// Do not print stack trace, a user-friendly error-message has already been logged
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} catch (IOException e) {
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