Move mpvcore/player/ to player/

1 files changed, 0 insertions, 471 deletions

diff --git a/mpvcore/player/audio.c b/mpvcore/player/audio.c
deleted file mode 100644
index ec2f039531..0000000000
--- a/mpvcore/player/audio.c
+++ /dev/null
@@ -1,471 +0,0 @@
-/*
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stddef.h>
-#include <stdbool.h>
-#include <inttypes.h>
-#include <math.h>
-#include <assert.h>
-
-#include "config.h"
-#include "talloc.h"
-
-#include "mpvcore/mp_msg.h"
-#include "mpvcore/options.h"
-#include "mpvcore/mp_common.h"
-
-#include "audio/mixer.h"
-#include "audio/audio.h"
-#include "audio/audio_buffer.h"
-#include "audio/decode/dec_audio.h"
-#include "audio/filter/af.h"
-#include "audio/out/ao.h"
-#include "demux/demux.h"
-#include "video/decode/dec_video.h"
-
-#include "mp_core.h"
-
-static int build_afilter_chain(struct MPContext *mpctx)
-{
-    struct dec_audio *d_audio = mpctx->d_audio;
-    struct ao *ao = mpctx->ao;
-    struct MPOpts *opts = mpctx->opts;
-
-    if (!d_audio)
-        return 0;
-
-    struct mp_audio in_format;
-    mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
-
-    int new_srate;
-    if (af_control_any_rev(d_audio->afilter, AF_CONTROL_SET_PLAYBACK_SPEED,
-                           &opts->playback_speed))
-        new_srate = in_format.rate;
-    else {
-        new_srate = in_format.rate * opts->playback_speed;
-        if (new_srate != ao->samplerate) {
-            // limits are taken from libaf/af_resample.c
-            if (new_srate < 8000)
-                new_srate = 8000;
-            if (new_srate > 192000)
-                new_srate = 192000;
-            opts->playback_speed = new_srate / (double)in_format.rate;
-        }
-    }
-    return audio_init_filters(d_audio, new_srate,
-                              &ao->samplerate, &ao->channels, &ao->format);
-}
-
-static int recreate_audio_filters(struct MPContext *mpctx)
-{
-    assert(mpctx->d_audio);
-
-    // init audio filters:
-    if (!build_afilter_chain(mpctx)) {
-        MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
-        return -1;
-    }
-
-    mixer_reinit_audio(mpctx->mixer, mpctx->ao, mpctx->d_audio->afilter);
-
-    return 0;
-}
-
-int reinit_audio_filters(struct MPContext *mpctx)
-{
-    struct dec_audio *d_audio = mpctx->d_audio;
-    if (!d_audio)
-        return -2;
-
-    af_uninit(mpctx->d_audio->afilter);
-    if (af_init(mpctx->d_audio->afilter) < 0)
-        return -1;
-    if (recreate_audio_filters(mpctx) < 0)
-        return -1;
-
-    return 0;
-}
-
-void reinit_audio_chain(struct MPContext *mpctx)
-{
-    struct MPOpts *opts = mpctx->opts;
-    struct sh_stream *sh = init_demux_stream(mpctx, STREAM_AUDIO);
-    if (!sh) {
-        uninit_player(mpctx, INITIALIZED_AO);
-        goto no_audio;
-    }
-
-    if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
-        mpctx->initialized_flags |= INITIALIZED_ACODEC;
-        assert(!mpctx->d_audio);
-        mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
-        mpctx->d_audio->opts = opts;
-        mpctx->d_audio->header = sh;
-        if (!audio_init_best_codec(mpctx->d_audio, opts->audio_decoders))
-            goto init_error;
-    }
-    assert(mpctx->d_audio);
-
-    struct mp_audio in_format;
-    mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
-
-    int ao_srate = opts->force_srate;
-    int ao_format = opts->audio_output_format;
-    struct mp_chmap ao_channels = {0};
-    if (mpctx->initialized_flags & INITIALIZED_AO) {
-        ao_srate    = mpctx->ao->samplerate;
-        ao_format   = mpctx->ao->format;
-        ao_channels = mpctx->ao->channels;
-    } else {
-        // Automatic downmix
-        if (mp_chmap_is_stereo(&opts->audio_output_channels) &&
-            !mp_chmap_is_stereo(&in_format.channels))
-        {
-            mp_chmap_from_channels(&ao_channels, 2);
-        }
-    }
-
-    // Determine what the filter chain outputs. build_afilter_chain() also
-    // needs this for testing whether playback speed is changed by resampling
-    // or using a special filter.
-    if (!audio_init_filters(mpctx->d_audio,  // preliminary init
-                            // input:
-                            in_format.rate,
-                            // output:
-                            &ao_srate, &ao_channels, &ao_format)) {
-        MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
-        goto init_error;
-    }
-
-    if (!(mpctx->initialized_flags & INITIALIZED_AO)) {
-        mpctx->initialized_flags |= INITIALIZED_AO;
-        mp_chmap_remove_useless_channels(&ao_channels,
-                                         &opts->audio_output_channels);
-        mpctx->ao = ao_init_best(mpctx->global, mpctx->input,
-                                 mpctx->encode_lavc_ctx, ao_srate, ao_format,
-                                 ao_channels);
-        struct ao *ao = mpctx->ao;
-        if (!ao) {
-            MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
-            goto init_error;
-        }
-
-        ao->buffer = mp_audio_buffer_create(ao);
-        mp_audio_buffer_reinit_fmt(ao->buffer, ao->format, &ao->channels,
-                                   ao->samplerate);
-
-        char *s = mp_audio_fmt_to_str(ao->samplerate, &ao->channels, ao->format);
-        MP_INFO(mpctx, "AO: [%s] %s\n", ao->driver->name, s);
-        talloc_free(s);
-        MP_VERBOSE(mpctx, "AO: Description: %s\n", ao->driver->description);
-        update_window_title(mpctx, true);
-    }
-
-    if (recreate_audio_filters(mpctx) < 0)
-        goto init_error;
-
-    mpctx->syncing_audio = true;
-    return;
-
-init_error:
-    uninit_player(mpctx, INITIALIZED_ACODEC | INITIALIZED_AO);
-    cleanup_demux_stream(mpctx, STREAM_AUDIO);
-no_audio:
-    mpctx->current_track[STREAM_AUDIO] = NULL;
-    MP_INFO(mpctx, "Audio: no audio\n");
-}
-
-// Return pts value corresponding to the end point of audio written to the
-// ao so far.
-double written_audio_pts(struct MPContext *mpctx)
-{
-    struct dec_audio *d_audio = mpctx->d_audio;
-    if (!d_audio)
-        return MP_NOPTS_VALUE;
-
-    struct mp_audio in_format;
-    mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
-
-    // first calculate the end pts of audio that has been output by decoder
-    double a_pts = d_audio->pts;
-    if (a_pts == MP_NOPTS_VALUE)
-        return MP_NOPTS_VALUE;
-
-    // d_audio->pts is the timestamp of the latest input packet with
-    // known pts that the decoder has decoded. d_audio->pts_bytes is
-    // the amount of bytes the decoder has written after that timestamp.
-    a_pts += d_audio->pts_offset / (double)in_format.rate;
-
-    // Now a_pts hopefully holds the pts for end of audio from decoder.
-    // Subtract data in buffers between decoder and audio out.
-
-    // Decoded but not filtered
-    a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
-
-    // Data buffered in audio filters, measured in seconds of "missing" output
-    double buffered_output = af_calc_delay(d_audio->afilter);
-
-    // Data that was ready for ao but was buffered because ao didn't fully
-    // accept everything to internal buffers yet
-    buffered_output += mp_audio_buffer_seconds(mpctx->ao->buffer);
-
-    // Filters divide audio length by playback_speed, so multiply by it
-    // to get the length in original units without speedup or slowdown
-    a_pts -= buffered_output * mpctx->opts->playback_speed;
-
-    return a_pts + mpctx->video_offset;
-}
-
-// Return pts value corresponding to currently playing audio.
-double playing_audio_pts(struct MPContext *mpctx)
-{
-    double pts = written_audio_pts(mpctx);
-    if (pts == MP_NOPTS_VALUE)
-        return pts;
-    return pts - mpctx->opts->playback_speed * ao_get_delay(mpctx->ao);
-}
-
-static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags,
-                       double pts)
-{
-    if (mpctx->paused)
-        return 0;
-    struct ao *ao = mpctx->ao;
-    ao->pts = pts;
-    double real_samplerate = ao->samplerate / mpctx->opts->playback_speed;
-    int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
-    assert(played <= data->samples);
-    if (played > 0) {
-        mpctx->shown_aframes += played;
-        mpctx->delay += played / real_samplerate;
-        // Keep correct pts for remaining data - could be used to flush
-        // remaining buffer when closing ao.
-        ao->pts += played / real_samplerate;
-        return played;
-    }
-    return 0;
-}
-
-static int write_silence_to_ao(struct MPContext *mpctx, int samples, int flags,
-                               double pts)
-{
-    struct mp_audio tmp = {0};
-    mp_audio_buffer_get_format(mpctx->ao->buffer, &tmp);
-    tmp.samples = samples;
-    char *p = talloc_size(NULL, tmp.samples * tmp.sstride);
-    for (int n = 0; n < tmp.num_planes; n++)
-        tmp.planes[n] = p;
-    mp_audio_fill_silence(&tmp, 0, tmp.samples);
-    int r = write_to_ao(mpctx, &tmp, 0, pts);
-    talloc_free(p);
-    return r;
-}
-
-#define ASYNC_PLAY_DONE -3
-static int audio_start_sync(struct MPContext *mpctx, int playsize)
-{
-    struct ao *ao = mpctx->ao;
-    struct MPOpts *opts = mpctx->opts;
-    struct dec_audio *d_audio = mpctx->d_audio;
-    int res;
-
-    assert(d_audio);
-
-    // Timing info may not be set without
-    res = audio_decode(d_audio, ao->buffer, 1);
-    if (res < 0)
-        return res;
-
-    int samples;
-    bool did_retry = false;
-    double written_pts;
-    double real_samplerate = ao->samplerate / opts->playback_speed;
-    bool hrseek = mpctx->hrseek_active;   // audio only hrseek
-    mpctx->hrseek_active = false;
-    while (1) {
-        written_pts = written_audio_pts(mpctx);
-        double ptsdiff;
-        if (hrseek)
-            ptsdiff = written_pts - mpctx->hrseek_pts;
-        else
-            ptsdiff = written_pts - mpctx->video_next_pts - mpctx->delay
-                      - mpctx->audio_delay;
-        samples = ptsdiff * real_samplerate;
-
-        // ogg demuxers give packets without timing
-        if (written_pts <= 1 && d_audio->pts == MP_NOPTS_VALUE) {
-            if (!did_retry) {
-                // Try to read more data to see packets that have pts
-                res = audio_decode(d_audio, ao->buffer, ao->samplerate);
-                if (res < 0)
-                    return res;
-                did_retry = true;
-                continue;
-            }
-            samples = 0;
-        }
-
-        if (fabs(ptsdiff) > 300 || isnan(ptsdiff))   // pts reset or just broken?
-            samples = 0;
-
-        if (samples > 0)
-            break;
-
-        mpctx->syncing_audio = false;
-        int skip_samples = -samples;
-        int a = MPMIN(skip_samples, MPMAX(playsize, 2500));
-        res = audio_decode(d_audio, ao->buffer, a);
-        if (skip_samples <= mp_audio_buffer_samples(ao->buffer)) {
-            mp_audio_buffer_skip(ao->buffer, skip_samples);
-            if (res < 0)
-                return res;
-            return audio_decode(d_audio, ao->buffer, playsize);
-        }
-        mp_audio_buffer_clear(ao->buffer);
-        if (res < 0)
-            return res;
-    }
-    if (hrseek)
-        // Don't add silence in audio-only case even if position is too late
-        return 0;
-    if (samples >= playsize) {
-        /* This case could fall back to the one below with
-         * samples = playsize, but then silence would keep accumulating
-         * in ao->buffer if the AO accepts less data than it asks for
-         * in playsize. */
-        write_silence_to_ao(mpctx, playsize, 0,
-                            written_pts - samples / real_samplerate);
-        return ASYNC_PLAY_DONE;
-    }
-    mpctx->syncing_audio = false;
-    mp_audio_buffer_prepend_silence(ao->buffer, samples);
-    return audio_decode(d_audio, ao->buffer, playsize);
-}
-
-int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
-{
-    struct MPOpts *opts = mpctx->opts;
-    struct ao *ao = mpctx->ao;
-    int playsize;
-    int playflags = 0;
-    bool audio_eof = false;
-    bool signal_eof = false;
-    bool partial_fill = false;
-    struct dec_audio *d_audio = mpctx->d_audio;
-    // Can't adjust the start of audio with spdif pass-through.
-    bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK);
-
-    assert(d_audio);
-
-    if (mpctx->paused)
-        playsize = 1;   // just initialize things (audio pts at least)
-    else
-        playsize = ao_get_space(ao);
-
-    // Coming here with hrseek_active still set means audio-only
-    if (!mpctx->d_video || !mpctx->sync_audio_to_video)
-        mpctx->syncing_audio = false;
-    if (!opts->initial_audio_sync || !modifiable_audio_format) {
-        mpctx->syncing_audio = false;
-        mpctx->hrseek_active = false;
-    }
-
-    int res;
-    if (mpctx->syncing_audio || mpctx->hrseek_active)
-        res = audio_start_sync(mpctx, playsize);
-    else
-        res = audio_decode(d_audio, ao->buffer, playsize);
-
-    if (res < 0) {  // EOF, error or format change
-        if (res == -2) {
-            /* The format change isn't handled too gracefully. A more precise
-             * implementation would require draining buffered old-format audio
-             * while displaying video, then doing the output format switch.
-             */
-            if (!mpctx->opts->gapless_audio)
-                uninit_player(mpctx, INITIALIZED_AO);
-            reinit_audio_chain(mpctx);
-            return -1;
-        } else if (res == ASYNC_PLAY_DONE)
-            return 0;
-        else if (demux_stream_eof(d_audio->header))
-            audio_eof = true;
-    }
-
-    if (endpts != MP_NOPTS_VALUE) {
-        double samples = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay)
-                         * ao->samplerate / opts->playback_speed;
-        if (playsize > samples) {
-            playsize = MPMAX(samples, 0);
-            audio_eof = true;
-            partial_fill = true;
-        }
-    }
-
-    if (playsize > mp_audio_buffer_samples(ao->buffer)) {
-        playsize = mp_audio_buffer_samples(ao->buffer);
-        partial_fill = true;
-    }
-    if (!playsize)
-        return partial_fill && audio_eof ? -2 : -partial_fill;
-
-    if (audio_eof && partial_fill) {
-        if (opts->gapless_audio) {
-            // With gapless audio, delay this to ao_uninit. There must be only
-            // 1 final chunk, and that is handled when calling ao_uninit().
-            signal_eof = true;
-        } else {
-            playflags |= AOPLAY_FINAL_CHUNK;
-        }
-    }
-
-    assert(ao->buffer_playable_samples <= mp_audio_buffer_samples(ao->buffer));
-
-    struct mp_audio data;
-    mp_audio_buffer_peek(ao->buffer, &data);
-    data.samples = MPMIN(data.samples, playsize);
-    int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx));
-    ao->buffer_playable_samples = playsize - played;
-
-    if (played > 0) {
-        mp_audio_buffer_skip(ao->buffer, played);
-    } else if (!mpctx->paused && audio_eof && ao_get_delay(ao) < .04) {
-        // Sanity check to avoid hanging in case current ao doesn't output
-        // partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
-        signal_eof = true;
-    }
-
-    return signal_eof ? -2 : -partial_fill;
-}
-
-// Drop data queued for output, or which the AO is currently outputting.
-void clear_audio_output_buffers(struct MPContext *mpctx)
-{
-    if (mpctx->ao) {
-        ao_reset(mpctx->ao);
-        mp_audio_buffer_clear(mpctx->ao->buffer);
-        mpctx->ao->buffer_playable_samples = 0;
-    }
-}
-
-// Drop decoded data queued for filtering.
-void clear_audio_decode_buffers(struct MPContext *mpctx)
-{
-    if (mpctx->d_audio)
-        mp_audio_buffer_clear(mpctx->d_audio->decode_buffer);
-}