Move mpvcore/player/ to player/

1 files changed, 471 insertions, 0 deletions

diff --git a/player/audio.c b/player/audio.c
new file mode 100644
index 0000000000..ec2f039531
--- /dev/null
+++ b/player/audio.c
@@ -0,0 +1,471 @@
+/*
+ * 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);
+}