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authorwm4 <wm4@nowhere>2018-01-29 06:18:33 +0100
committerKevin Mitchell <kevmitch@gmail.com>2018-01-30 03:10:27 -0800
commit76e7e78ce989aad546310b9808cf7f96f23a281f (patch)
treee4432df6f925d2a1f9e5022373d359b626d9ea09 /player
parent054c02ad64a62dd8393bde0fd73edeaa71048722 (diff)
downloadmpv-76e7e78ce989aad546310b9808cf7f96f23a281f.tar.bz2
mpv-76e7e78ce989aad546310b9808cf7f96f23a281f.tar.xz
audio: move to decoder wrapper
Use the decoder wrapper that was introduced for video. This removes all code duplication the old audio decoder wrapper had with the video code. (The audio wrapper was copy pasted from the video one over a decade ago, and has been kept in sync ever since by the power of copy&paste. Since the original copy&paste was possibly done by someone who did not answer to the LGPL relicensing, this should also remove all doubts about whether any of this code is left, since we now completely remove any code that could possibly have been based on it.) There is some complication with spdif handling, and a minor behavior change (it will restrict the list of codecs to spdif if spdif is to be used), but there should not be any difference in practice.
Diffstat (limited to 'player')
-rw-r--r--player/audio.c220
-rw-r--r--player/command.c5
-rw-r--r--player/core.h10
-rw-r--r--player/loadfile.c30
-rw-r--r--player/main.c1
-rw-r--r--player/playloop.c38
-rw-r--r--player/video.c1
7 files changed, 75 insertions, 230 deletions
diff --git a/player/audio.c b/player/audio.c
index ab53ab3b86..5b061efca1 100644
--- a/player/audio.c
+++ b/player/audio.c
@@ -33,21 +33,17 @@
#include "audio/audio_buffer.h"
#include "audio/format.h"
-#include "audio/decode/dec_audio.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
+#include "filters/f_decoder_wrapper.h"
#include "core.h"
#include "command.h"
enum {
AD_OK = 0,
- AD_ERR = -1,
AD_EOF = -2,
- AD_NEW_FMT = -3,
AD_WAIT = -4,
- AD_NO_PROGRESS = -5,
- AD_STARVE = -6,
};
// Try to reuse the existing filters to change playback speed. If it works,
@@ -183,17 +179,11 @@ void update_playback_speed(struct MPContext *mpctx)
static void ao_chain_reset_state(struct ao_chain *ao_c)
{
ao_c->last_out_pts = MP_NOPTS_VALUE;
- ao_c->pts = MP_NOPTS_VALUE;
ao_c->pts_reset = false;
- TA_FREEP(&ao_c->input_frame);
TA_FREEP(&ao_c->output_frame);
+ ao_c->out_eof = false;
mp_audio_buffer_clear(ao_c->ao_buffer);
-
- if (ao_c->audio_src)
- audio_reset_decoding(ao_c->audio_src);
-
- ao_c->filter_src_got_eof = false;
}
void reset_audio_state(struct MPContext *mpctx)
@@ -226,16 +216,16 @@ static void ao_chain_uninit(struct ao_chain *ao_c)
if (track) {
assert(track->ao_c == ao_c);
track->ao_c = NULL;
- assert(track->d_audio == ao_c->audio_src);
- track->d_audio = NULL;
- audio_uninit(ao_c->audio_src);
+ if (ao_c->dec_src)
+ assert(track->dec->f->pins[0] == ao_c->dec_src);
+ talloc_free(track->dec->f);
+ track->dec = NULL;
}
if (ao_c->filter_src)
mp_pin_disconnect(ao_c->filter_src);
talloc_free(ao_c->filter->f);
- talloc_free(ao_c->input_frame);
talloc_free(ao_c->output_frame);
talloc_free(ao_c->ao_buffer);
talloc_free(ao_c);
@@ -361,12 +351,12 @@ static void reinit_audio_filters_and_output(struct MPContext *mpctx)
if (!mpctx->ao) {
// If spdif was used, try to fallback to PCM.
- if (spdif_fallback && ao_c->audio_src) {
+ if (spdif_fallback && ao_c->track && ao_c->track->dec) {
MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
ao_c->spdif_passthrough = false;
ao_c->spdif_failed = true;
- ao_c->audio_src->try_spdif = false;
- if (!audio_init_best_codec(ao_c->audio_src))
+ ao_c->track->dec->try_spdif = false;
+ if (!mp_decoder_wrapper_reinit(ao_c->track->dec))
goto init_error;
reset_audio_state(mpctx);
mp_output_chain_reset_harder(ao_c->filter);
@@ -408,21 +398,18 @@ init_error:
int init_audio_decoder(struct MPContext *mpctx, struct track *track)
{
- assert(!track->d_audio);
+ assert(!track->dec);
if (!track->stream)
goto init_error;
- track->d_audio = talloc_zero(NULL, struct dec_audio);
- struct dec_audio *d_audio = track->d_audio;
- d_audio->log = mp_log_new(d_audio, mpctx->log, "!ad");
- d_audio->global = mpctx->global;
- d_audio->opts = mpctx->opts;
- d_audio->header = track->stream;
- d_audio->codec = track->stream->codec;
+ track->dec = mp_decoder_wrapper_create(mpctx->filter_root, track->stream);
+ if (!track->dec)
+ goto init_error;
- d_audio->try_spdif = true;
+ if (track->ao_c)
+ track->dec->try_spdif = true;
- if (!audio_init_best_codec(d_audio))
+ if (!mp_decoder_wrapper_reinit(track->dec))
goto init_error;
return 1;
@@ -431,8 +418,6 @@ init_error:
if (track->sink)
mp_pin_disconnect(track->sink);
track->sink = NULL;
- audio_uninit(track->d_audio);
- track->d_audio = NULL;
error_on_track(mpctx, track);
return 0;
}
@@ -462,7 +447,7 @@ void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track)
ao_c->filter =
mp_output_chain_create(mpctx->filter_root, MP_OUTPUT_CHAIN_AUDIO);
ao_c->spdif_passthrough = true;
- ao_c->pts = MP_NOPTS_VALUE;
+ ao_c->last_out_pts = MP_NOPTS_VALUE;
ao_c->ao_buffer = mp_audio_buffer_create(NULL);
ao_c->ao = mpctx->ao;
@@ -471,7 +456,8 @@ void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track)
track->ao_c = ao_c;
if (!init_audio_decoder(mpctx, track))
goto init_error;
- ao_c->audio_src = track->d_audio;
+ ao_c->dec_src = track->dec->f->pins[0];
+ mp_pin_connect(ao_c->filter->f->pins[0], ao_c->dec_src);
}
reset_audio_state(mpctx);
@@ -643,7 +629,7 @@ static bool get_sync_samples(struct MPContext *mpctx, int *skip)
static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c,
- int minsamples, double endpts, bool eof, bool *seteof)
+ int minsamples, double endpts, bool *seteof)
{
struct mp_audio_buffer *outbuf = ao_c->ao_buffer;
@@ -671,16 +657,39 @@ static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c,
struct mp_frame frame = mp_pin_out_read(ao_c->filter->f->pins[1]);
if (frame.type == MP_FRAME_AUDIO) {
ao_c->output_frame = frame.data;
+ ao_c->out_eof = false;
+
+ double pts = mp_aframe_get_pts(ao_c->output_frame);
+ if (pts != MP_NOPTS_VALUE) {
+ // Attempt to detect jumps in PTS. Even for the lowest
+ // sample rates and with worst container rounded timestamp,
+ // this should be a margin more than enough.
+ double desync = pts - ao_c->last_out_pts;
+ if (ao_c->last_out_pts != MP_NOPTS_VALUE && fabs(desync) > 0.1)
+ {
+ MP_WARN(ao_c, "Invalid audio PTS: %f -> %f\n",
+ ao_c->last_out_pts, pts);
+ if (desync >= 5)
+ ao_c->pts_reset = true;
+ }
+ }
ao_c->last_out_pts = mp_aframe_end_pts(ao_c->output_frame);
} else if (frame.type == MP_FRAME_EOF) {
- *seteof = true;
+ ao_c->out_eof = true;
} else if (frame.type) {
MP_ERR(mpctx, "unknown frame type\n");
+ mp_frame_unref(&frame);
}
}
- if (!ao_c->output_frame)
- return false; // out of data
+ // out of data
+ if (!ao_c->output_frame) {
+ if (ao_c->out_eof) {
+ *seteof = true;
+ return true;
+ }
+ return false;
+ }
if (cursamples + mp_aframe_get_size(ao_c->output_frame) > maxsamples) {
if (cursamples < maxsamples) {
@@ -702,43 +711,6 @@ static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c,
return true;
}
-static int decode_new_frame(struct ao_chain *ao_c)
-{
- if (ao_c->input_frame)
- return AD_OK;
-
- int res = DATA_EOF;
- if (ao_c->filter_src) {
- struct mp_frame frame = mp_pin_out_read(ao_c->filter_src);
- if (frame.type == MP_FRAME_EOF) {
- res = DATA_EOF;
- ao_c->filter_src_got_eof = true;
- } else if (frame.type == MP_FRAME_AUDIO) {
- res = DATA_OK;
- ao_c->input_frame = frame.data;
- ao_c->filter_src_got_eof = false;
- } else if (frame.type) {
- MP_ERR(ao_c, "unexpected frame type\n");
- mp_frame_unref(&frame);
- res = DATA_EOF;
- } else {
- res = ao_c->filter_src_got_eof ? DATA_EOF : DATA_WAIT;
- }
- } else if (ao_c->audio_src) {
- audio_work(ao_c->audio_src);
- res = audio_get_frame(ao_c->audio_src, &ao_c->input_frame);
- }
-
- switch (res) {
- case DATA_OK: return AD_OK;
- case DATA_WAIT: return AD_WAIT;
- case DATA_AGAIN: return AD_NO_PROGRESS;
- case DATA_STARVE: return AD_STARVE;
- case DATA_EOF: return AD_EOF;
- default: abort();
- }
-}
-
/* Try to get at least minsamples decoded+filtered samples in outbuf
* (total length including possible existing data).
* Return 0 on success, or negative AD_* error code.
@@ -749,64 +721,12 @@ static int filter_audio(struct MPContext *mpctx, struct mp_audio_buffer *outbuf,
{
struct ao_chain *ao_c = mpctx->ao_chain;
- MP_STATS(ao_c, "start audio");
-
double endpts = get_play_end_pts(mpctx);
bool eof = false;
- int res;
- while (1) {
- res = 0;
-
- if (copy_output(mpctx, ao_c, minsamples, endpts, false, &eof))
- break;
-
- res = decode_new_frame(ao_c);
- if (res == AD_NO_PROGRESS)
- continue;
- if (res == AD_WAIT || res == AD_STARVE)
- break;
- if (res < 0) {
- // drain filters first (especially for true EOF case)
- if (!ao_c->filter->got_input_eof)
- mp_pin_in_write(ao_c->filter->f->pins[0], MP_EOF_FRAME);
- copy_output(mpctx, ao_c, minsamples, endpts, true, &eof);
- break;
- }
- assert(ao_c->input_frame);
-
- double pts = mp_aframe_get_pts(ao_c->input_frame);
- if (pts == MP_NOPTS_VALUE) {
- ao_c->pts = MP_NOPTS_VALUE;
- } else {
- // Attempt to detect jumps in PTS. Even for the lowest sample rates
- // and with worst container rounded timestamp, this should be a
- // margin more than enough.
- double desync = pts - ao_c->pts;
- if (ao_c->pts != MP_NOPTS_VALUE && fabs(desync) > 0.1) {
- MP_WARN(ao_c, "Invalid audio PTS: %f -> %f\n",
- ao_c->pts, pts);
- if (desync >= 5)
- ao_c->pts_reset = true;
- }
- ao_c->pts = mp_aframe_end_pts(ao_c->input_frame);
- }
-
- if (!mp_pin_in_needs_data(ao_c->filter->f->pins[0])) {
- res = AD_WAIT;
- break;
- }
- mp_pin_in_write(ao_c->filter->f->pins[0],
- MAKE_FRAME(MP_FRAME_AUDIO, ao_c->input_frame));
- ao_c->input_frame = NULL;
- }
-
- if (res == 0 && mp_audio_buffer_samples(outbuf) < minsamples && eof)
- res = AD_EOF;
-
- MP_STATS(ao_c, "end audio");
-
- return res;
+ if (!copy_output(mpctx, ao_c, minsamples, endpts, &eof))
+ return AD_WAIT;
+ return eof ? AD_EOF : AD_OK;
}
void reload_audio_output(struct MPContext *mpctx)
@@ -818,17 +738,23 @@ void reload_audio_output(struct MPContext *mpctx)
uninit_audio_out(mpctx);
reinit_audio_filters(mpctx); // mostly to issue refresh seek
+ struct ao_chain *ao_c = mpctx->ao_chain;
+
+ if (ao_c) {
+ reset_audio_state(mpctx);
+ mp_output_chain_reset_harder(ao_c->filter);
+ }
+
// Whether we can use spdif might have changed. If we failed to use spdif
// in the previous initialization, try it with spdif again (we'll fallback
// to PCM again if necessary).
- struct ao_chain *ao_c = mpctx->ao_chain;
- if (ao_c) {
- struct dec_audio *d_audio = ao_c->audio_src;
- if (d_audio && ao_c->spdif_failed) {
+ if (ao_c && ao_c->track) {
+ struct mp_decoder_wrapper *dec = ao_c->track->dec;
+ if (dec && ao_c->spdif_failed) {
ao_c->spdif_passthrough = true;
ao_c->spdif_failed = false;
- d_audio->try_spdif = true;
- if (!audio_init_best_codec(d_audio)) {
+ dec->try_spdif = true;
+ if (!mp_decoder_wrapper_reinit(dec)) {
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, ao_c->track);
}
@@ -857,29 +783,13 @@ void fill_audio_out_buffers(struct MPContext *mpctx)
return;
}
- if (ao_c->input_frame && mp_pin_in_needs_data(ao_c->filter->f->pins[0])) {
- mp_pin_in_write(ao_c->filter->f->pins[0],
- MAKE_FRAME(MP_FRAME_AUDIO, ao_c->input_frame));
- ao_c->input_frame = NULL;
- }
-
// (if AO is set due to gapless from previous file, then we can try to
// filter normally until the filter tells us to change the AO)
if (!mpctx->ao) {
- mp_pin_out_request_data(ao_c->filter->f->pins[1]);
// Probe the initial audio format. Returns AD_OK (and does nothing) if
// the format is already known.
- int r = AD_NO_PROGRESS;
- while (r == AD_NO_PROGRESS)
- r = decode_new_frame(mpctx->ao_chain);
- if (r == AD_WAIT)
- return; // continue later when new data is available
- if (r == AD_EOF) {
- mpctx->audio_status = STATUS_EOF;
- return;
- }
+ mp_pin_out_request_data(ao_c->filter->f->pins[1]);
reinit_audio_filters_and_output(mpctx);
- mp_wakeup_core(mpctx);
return; // try again next iteration
}
@@ -949,12 +859,6 @@ void fill_audio_out_buffers(struct MPContext *mpctx)
}
if (status == AD_WAIT)
return;
- if (status == AD_NO_PROGRESS || status == AD_STARVE) {
- mp_wakeup_core(mpctx);
- return;
- }
- if (status == AD_ERR)
- mp_wakeup_core(mpctx);
working = true;
}
diff --git a/player/command.c b/player/command.c
index d1de5a86ff..1b074b8767 100644
--- a/player/command.c
+++ b/player/command.c
@@ -57,7 +57,6 @@
#include "audio/aframe.h"
#include "audio/format.h"
#include "audio/out/ao.h"
-#include "audio/decode/dec_audio.h"
#include "video/out/bitmap_packer.h"
#include "options/path.h"
#include "screenshot.h"
@@ -2038,7 +2037,7 @@ static int mp_property_audio_codec(void *ctx, struct m_property *prop,
{
MPContext *mpctx = ctx;
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
- const char *c = track && track->d_audio ? track->d_audio->decoder_desc : NULL;
+ const char *c = track && track->dec ? track->dec->decoder_desc : NULL;
return m_property_strdup_ro(action, arg, c);
}
@@ -2186,8 +2185,6 @@ static int get_track_entry(int item, int action, void *arg, void *ctx)
const char *decoder_desc = NULL;
if (track->dec)
decoder_desc = track->dec->decoder_desc;
- if (track->d_audio)
- decoder_desc = track->d_audio->decoder_desc;
bool has_rg = track->stream && track->stream->codec->replaygain_data;
struct replaygain_data rg = has_rg ? *track->stream->codec->replaygain_data
diff --git a/player/core.h b/player/core.h
index f27c30b145..8a77690de6 100644
--- a/player/core.h
+++ b/player/core.h
@@ -154,13 +154,11 @@ struct track {
// Current decoding state (NULL if selected==false)
struct mp_decoder_wrapper *dec;
- struct dec_audio *d_audio;
// Where the decoded result goes to (one of them is not NULL if active)
struct vo_chain *vo_c;
struct ao_chain *ao_c;
struct mp_pin *sink;
- bool sink_eof; // whether it got passed EOF
// For stream recording (remuxing mode).
struct mp_recorder_sink *remux_sink;
@@ -190,7 +188,6 @@ struct vo_chain {
struct ao_chain {
struct mp_log *log;
- double pts; // timestamp of first sample output by decoder
bool spdif_passthrough, spdif_failed;
bool pts_reset;
@@ -200,18 +197,15 @@ struct ao_chain {
struct mp_audio_buffer *ao_buffer;
double ao_resume_time;
- // 1-element input frame queue.
- struct mp_aframe *input_frame;
-
// 1-element output frame queue.
struct mp_aframe *output_frame;
+ bool out_eof;
double last_out_pts;
struct track *track;
struct mp_pin *filter_src;
- bool filter_src_got_eof; // whether this returned EOF last time
- struct dec_audio *audio_src;
+ struct mp_pin *dec_src;
};
/* Note that playback can be paused, stopped, etc. at any time. While paused,
diff --git a/player/loadfile.c b/player/loadfile.c
index d35ae6ad6b..44f0f970ac 100644
--- a/player/loadfile.c
+++ b/player/loadfile.c
@@ -43,7 +43,6 @@
#include "common/recorder.h"
#include "input/input.h"
-#include "audio/decode/dec_audio.h"
#include "audio/out/ao.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/f_lavfi.h"
@@ -984,14 +983,10 @@ static void cleanup_deassociated_complex_filters(struct MPContext *mpctx)
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
if (!(track->sink || track->vo_c || track->ao_c)) {
- if (track->dec && !track->vo_c) {
+ if (track->dec && !track->vo_c && !track->ao_c) {
talloc_free(track->dec->f);
track->dec->f = NULL;
}
- if (track->d_audio && !track->ao_c) {
- audio_uninit(track->d_audio);
- track->d_audio = NULL;
- }
track->selected = false;
}
}
@@ -1001,7 +996,7 @@ static void cleanup_deassociated_complex_filters(struct MPContext *mpctx)
{
uninit_video_chain(mpctx);
}
- if (mpctx->ao_chain && !mpctx->ao_chain->audio_src &&
+ if (mpctx->ao_chain && !mpctx->ao_chain->dec_src &&
!mpctx->ao_chain->filter_src)
{
uninit_audio_chain(mpctx);
@@ -1094,17 +1089,16 @@ static int reinit_complex_filters(struct MPContext *mpctx, bool force_uninit)
pad = mp_filter_get_named_pin(mpctx->lavfi, "ao");
if (pad && mp_pin_get_dir(pad) == MP_PIN_OUT) {
if (mpctx->ao_chain) {
- if (mpctx->ao_chain->audio_src) {
- MP_ERR(mpctx, "Pad ao tries to connect to already used AO.\n");
- goto done;
- }
+ MP_ERR(mpctx, "Pad ao tries to connect to already used AO.\n");
+ goto done;
} else {
reinit_audio_chain_src(mpctx, NULL);
if (!mpctx->ao_chain)
goto done;
}
- mp_pin_set_manual_connection(pad, true);
- mpctx->ao_chain->filter_src = pad;
+ struct ao_chain *ao_c = mpctx->ao_chain;
+ ao_c->filter_src = pad;
+ mp_pin_connect(ao_c->filter->f->pins[0], ao_c->filter_src);
}
for (int n = 0; n < mpctx->num_tracks; n++) {
@@ -1115,8 +1109,9 @@ static int reinit_complex_filters(struct MPContext *mpctx, bool force_uninit)
mp_pin_connect(track->sink, track->dec->f->pins[0]);
}
if (track->sink && track->type == STREAM_AUDIO) {
- if (!track->d_audio && !init_audio_decoder(mpctx, track))
+ if (!track->dec && !init_audio_decoder(mpctx, track))
goto done;
+ mp_pin_connect(track->sink, track->dec->f->pins[0]);
}
}
@@ -1588,8 +1583,6 @@ static void set_track_recorder_sink(struct track *track,
sub_set_recorder_sink(track->d_sub, sink);
if (track->dec)
track->dec->recorder_sink = sink;
- if (track->d_audio)
- track->d_audio->recorder_sink = sink;
track->remux_sink = sink;
}
@@ -1631,11 +1624,8 @@ void open_recorder(struct MPContext *mpctx, bool on_init)
for (int n = 0; n < mpctx->num_tracks; n++) {
struct track *track = mpctx->tracks[n];
- if (track->stream && track->selected &&
- (track->d_sub || track->dec || track->d_audio))
- {
+ if (track->stream && track->selected && (track->d_sub || track->dec))
MP_TARRAY_APPEND(NULL, streams, num_streams, track->stream);
- }
}
mpctx->recorder = mp_recorder_create(mpctx->global, mpctx->opts->record_file,
diff --git a/player/main.c b/player/main.c
index 98abbc8e4f..711b413735 100644
--- a/player/main.c
+++ b/player/main.c
@@ -50,7 +50,6 @@
#include "options/path.h"
#include "input/input.h"
-#include "audio/decode/dec_audio.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "stream/stream.h"
diff --git a/player/playloop.c b/player/playloop.c
index 748469354d..e77200f2d7 100644
--- a/player/playloop.c
+++ b/player/playloop.c
@@ -39,7 +39,6 @@
#include "osdep/terminal.h"
#include "osdep/timer.h"
-#include "audio/decode/dec_audio.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "stream/stream.h"
@@ -212,12 +211,6 @@ void add_step_frame(struct MPContext *mpctx, int dir)
// Clear some playback-related fields on file loading or after seeks.
void reset_playback_state(struct MPContext *mpctx)
{
- for (int n = 0; n < mpctx->num_tracks; n++) {
- if (mpctx->tracks[n]->d_audio)
- audio_reset_decoding(mpctx->tracks[n]->d_audio);
- mpctx->tracks[n]->sink_eof = false;
- }
-
mp_filter_reset(mpctx->filter_root);
reset_video_state(mpctx);
@@ -1076,35 +1069,6 @@ static void handle_eof(struct MPContext *mpctx)
}
}
-static void handle_complex_filter_decoders(struct MPContext *mpctx)
-{
- if (!mpctx->lavfi)
- return;
-
- for (int n = 0; n < mpctx->num_tracks; n++) {
- struct track *track = mpctx->tracks[n];
- if (!track->selected)
- continue;
- if (track->d_audio) {
- if (!track->sink || !mp_pin_in_needs_data(track->sink))
- continue;
- audio_work(track->d_audio);
- struct mp_aframe *fr;
- int res = audio_get_frame(track->d_audio, &fr);
- if (res == DATA_OK) {
- mp_pin_in_write(track->sink, MAKE_FRAME(MP_FRAME_AUDIO, fr));
- track->sink_eof = false;
- } else if (res == DATA_EOF) {
- if (!track->sink_eof)
- mp_pin_in_write(track->sink, MP_EOF_FRAME);
- track->sink_eof = true;
- } else if (res == DATA_AGAIN) {
- mp_wakeup_core(mpctx);
- }
- }
- }
-}
-
void run_playloop(struct MPContext *mpctx)
{
#if HAVE_ENCODING
@@ -1116,8 +1080,6 @@ void run_playloop(struct MPContext *mpctx)
update_demuxer_properties(mpctx);
- handle_complex_filter_decoders(mpctx);
-
handle_cursor_autohide(mpctx);
handle_vo_events(mpctx);
handle_command_updates(mpctx);
diff --git a/player/video.c b/player/video.c
index 48b02ecec7..619c73e3f1 100644
--- a/player/video.c
+++ b/player/video.c
@@ -41,7 +41,6 @@
#include "video/hwdec.h"
#include "filters/f_decoder_wrapper.h"
#include "video/out/vo.h"
-#include "audio/decode/dec_audio.h"
#include "core.h"
#include "command.h"