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🎥 Command line media player
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#include <stddef.h>#include <stdbool.h>#include <inttypes.h>#include <limits.h>#include <math.h>#include <assert.h>
#include "config.h"#include "mpv_talloc.h"
#include "common/msg.h"#include "common/encode.h"#include "options/options.h"#include "common/common.h"#include "osdep/timer.h"
#include "audio/audio_buffer.h"#include "audio/aconverter.h"#include "audio/format.h"#include "audio/decode/dec_audio.h"#include "audio/out/ao.h"#include "demux/demux.h"#include "video/decode/dec_video.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,};
#if HAVE_LIBAF
#include "audio/audio.h"#include "audio/filter/af.h"
// Use pitch correction only for speed adjustments by the user, not minor sync// correction ones.static int get_speed_method(struct MPContext *mpctx){ return mpctx->opts->pitch_correction && mpctx->opts->playback_speed != 1.0 ? AF_CONTROL_SET_PLAYBACK_SPEED : AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;}
// Try to reuse the existing filters to change playback speed. If it works,// return true; if filter recreation is needed, return false.static bool update_speed_filters(struct MPContext *mpctx){ struct af_stream *afs = mpctx->ao_chain->af; double speed = mpctx->audio_speed;
if (afs->initialized < 1) return false;
// Make sure only exactly one filter changes speed; resetting them all // and setting 1 filter is the easiest way to achieve this. af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &(double){1}); af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &(double){1});
if (speed == 1.0) return !af_find_by_label(afs, "playback-speed");
// Compatibility: if the user uses --af=scaletempo, always use this // filter to change speed. Don't insert a second filter (any) either. if (!af_find_by_label(afs, "playback-speed") && af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed)) return true;
return !!af_control_any_rev(afs, get_speed_method(mpctx), &speed);}
// Update speed, and insert/remove filters if necessary.static void recreate_speed_filters(struct MPContext *mpctx){ struct af_stream *afs = mpctx->ao_chain->af;
if (update_speed_filters(mpctx)) return;
if (af_remove_by_label(afs, "playback-speed") < 0) goto fail;
if (mpctx->audio_speed == 1.0) return;
int method = get_speed_method(mpctx); char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED ? "scaletempo" : "lavrresample";
if (!af_add(afs, filter, "playback-speed", NULL)) goto fail;
if (!update_speed_filters(mpctx)) goto fail;
return;
fail: mpctx->opts->playback_speed = 1.0; mpctx->speed_factor_a = 1.0; mpctx->audio_speed = 1.0; mp_notify(mpctx, MP_EVENT_CHANGE_ALL, NULL);}
static int recreate_audio_filters(struct MPContext *mpctx){ assert(mpctx->ao_chain);
struct af_stream *afs = mpctx->ao_chain->af; if (afs->initialized < 1 && af_init(afs) < 0) goto fail;
recreate_speed_filters(mpctx); if (afs->initialized < 1 && af_init(afs) < 0) goto fail;
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return 0;
fail: MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n"); return -1;}
int reinit_audio_filters(struct MPContext *mpctx){ struct ao_chain *ao_c = mpctx->ao_chain; if (!ao_c) return 0;
double delay = 0; if (ao_c->af->initialized > 0) delay = af_calc_delay(ao_c->af);
af_uninit(ao_c->af); if (recreate_audio_filters(mpctx) < 0) return -1;
// Only force refresh if the amount of dropped buffered data is going to // cause "issues" for the A/V sync logic. if (mpctx->audio_status == STATUS_PLAYING && delay > 0.2) issue_refresh_seek(mpctx, MPSEEK_EXACT); return 1;}
#else /* HAVE_LIBAV */
int reinit_audio_filters(struct MPContext *mpctx) { return 0; }
#endif /* else HAVE_LIBAF */
static double db_gain(double db){ return pow(10.0, db/20.0);}
static float compute_replaygain(struct MPContext *mpctx){ struct MPOpts *opts = mpctx->opts;
float rgain = 1.0;
struct replaygain_data *rg = NULL; struct track *track = mpctx->current_track[0][STREAM_AUDIO]; if (track) rg = track->stream->codec->replaygain_data; if (opts->rgain_mode && rg) { MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n", rg->track_gain, rg->track_peak, rg->album_gain, rg->album_peak);
float gain, peak; if (opts->rgain_mode == 1) { gain = rg->track_gain; peak = rg->track_peak; } else { gain = rg->album_gain; peak = rg->album_peak; }
gain += opts->rgain_preamp; rgain = db_gain(gain);
MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);
if (!opts->rgain_clip) { // clipping prevention rgain = MPMIN(rgain, 1.0 / peak); MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain); } } else if (opts->rgain_fallback) { rgain = db_gain(opts->rgain_fallback); MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain); }
return rgain;}
// Called when opts->softvol_volume or opts->softvol_mute were changed.void audio_update_volume(struct MPContext *mpctx){ struct MPOpts *opts = mpctx->opts; struct ao_chain *ao_c = mpctx->ao_chain; if (!ao_c || !ao_c->ao) return;
float gain = MPMAX(opts->softvol_volume / 100.0, 0); gain = pow(gain, 3); gain *= compute_replaygain(mpctx); if (opts->softvol_mute == 1) gain = 0.0;
ao_set_gain(ao_c->ao, gain);}
// Call this if opts->playback_speed or mpctx->speed_factor_* change.void update_playback_speed(struct MPContext *mpctx){ mpctx->audio_speed = mpctx->opts->playback_speed * mpctx->speed_factor_a; mpctx->video_speed = mpctx->opts->playback_speed * mpctx->speed_factor_v;
#if HAVE_LIBAF if (!mpctx->ao_chain || mpctx->ao_chain->af->initialized < 1) return;
if (!update_speed_filters(mpctx)) recreate_audio_filters(mpctx);#endif}
static void ao_chain_reset_state(struct ao_chain *ao_c){ ao_c->pts = MP_NOPTS_VALUE; ao_c->pts_reset = false; TA_FREEP(&ao_c->input_frame); TA_FREEP(&ao_c->output_frame);#if HAVE_LIBAF af_seek_reset(ao_c->af);#endif if (ao_c->conv) mp_aconverter_flush(ao_c->conv); mp_audio_buffer_clear(ao_c->ao_buffer);
if (ao_c->audio_src) audio_reset_decoding(ao_c->audio_src);}
void reset_audio_state(struct MPContext *mpctx){ if (mpctx->ao_chain) ao_chain_reset_state(mpctx->ao_chain); mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF; mpctx->delay = 0; mpctx->audio_drop_throttle = 0; mpctx->audio_stat_start = 0; mpctx->audio_allow_second_chance_seek = false;}
void uninit_audio_out(struct MPContext *mpctx){ if (mpctx->ao) { // Note: with gapless_audio, stop_play is not correctly set if (mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE) ao_drain(mpctx->ao); ao_uninit(mpctx->ao);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL); } mpctx->ao = NULL; talloc_free(mpctx->ao_decoder_fmt); mpctx->ao_decoder_fmt = NULL;}
static void ao_chain_uninit(struct ao_chain *ao_c){ struct track *track = ao_c->track; 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->filter_src) lavfi_set_connected(ao_c->filter_src, false);
#if HAVE_LIBAF af_destroy(ao_c->af);#endif talloc_free(ao_c->conv); talloc_free(ao_c->input_frame); talloc_free(ao_c->input_format); talloc_free(ao_c->output_frame); talloc_free(ao_c->filter_input_format); talloc_free(ao_c->ao_buffer); talloc_free(ao_c);}
void uninit_audio_chain(struct MPContext *mpctx){ if (mpctx->ao_chain) { ao_chain_uninit(mpctx->ao_chain); mpctx->ao_chain = NULL;
mpctx->audio_status = STATUS_EOF;
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL); }}
static char *audio_config_to_str_buf(char *buf, size_t buf_sz, int rate, int format, struct mp_chmap channels){ char ch[128]; mp_chmap_to_str_buf(ch, sizeof(ch), &channels); char *hr_ch = mp_chmap_to_str_hr(&channels); if (strcmp(hr_ch, ch) != 0) mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch); snprintf(buf, buf_sz, "%dHz %s %dch %s", rate, ch, channels.num, af_fmt_to_str(format)); return buf;}
static void reinit_audio_filters_and_output(struct MPContext *mpctx){ struct MPOpts *opts = mpctx->opts; struct ao_chain *ao_c = mpctx->ao_chain; assert(ao_c); struct track *track = ao_c->track;
if (!mp_aframe_config_is_valid(ao_c->input_format)) { // We don't know the audio format yet - so configure it later as we're // resyncing. fill_audio_buffers() will call this function again. mp_wakeup_core(mpctx); return; }
// Weak gapless audio: drain AO on decoder format changes if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 && !mp_aframe_config_equals(mpctx->ao_decoder_fmt, ao_c->input_format)) { uninit_audio_out(mpctx); }
TA_FREEP(&ao_c->output_frame);
int out_rate = 0; int out_format = 0; struct mp_chmap out_channels = {0}; if (mpctx->ao) { ao_get_format(mpctx->ao, &out_rate, &out_format, &out_channels); } else if (af_fmt_is_pcm(mp_aframe_get_format(ao_c->input_format))) { out_rate = opts->force_srate; out_format = opts->audio_output_format; if (opts->audio_output_channels.num_chmaps == 1) out_channels = opts->audio_output_channels.chmaps[0]; }
#if HAVE_LIBAF struct af_stream *afs = ao_c->af;
struct mp_audio in_format; mp_audio_config_from_aframe(&in_format, ao_c->input_format); if (mpctx->ao && mp_audio_config_equals(&in_format, &afs->input)) return;
afs->output = (struct mp_audio){0}; afs->output.rate = out_rate; mp_audio_set_format(&afs->output, out_format); mp_audio_set_channels(&afs->output, &out_channels);
// filter input format: same as codec's output format: afs->input = in_format;
// Determine what the filter chain outputs. recreate_audio_filters() also // needs this for testing whether playback speed is changed by resampling // or using a special filter. if (af_init(afs) < 0) { MP_ERR(mpctx, "Error at audio filter chain pre-init!\n"); goto init_error; }
out_rate = afs->output.rate; out_format = afs->output.format; out_channels = afs->output.channels;#else if (mpctx->ao && ao_c->filter_input_format && mp_aframe_config_equals(ao_c->filter_input_format, ao_c->input_format)) return;
TA_FREEP(&ao_c->filter_input_format);
if (!out_rate) out_rate = mp_aframe_get_rate(ao_c->input_format); if (!out_format) out_format = mp_aframe_get_format(ao_c->input_format); if (!out_channels.num) mp_aframe_get_chmap(ao_c->input_format, &out_channels);#endif
if (!mpctx->ao) { int ao_flags = 0; bool spdif_fallback = af_fmt_is_spdif(out_format) && ao_c->spdif_passthrough;
if (opts->ao_null_fallback && !spdif_fallback) ao_flags |= AO_INIT_NULL_FALLBACK;
if (opts->audio_stream_silence) ao_flags |= AO_INIT_STREAM_SILENCE;
if (opts->audio_exclusive) ao_flags |= AO_INIT_EXCLUSIVE;
if (af_fmt_is_pcm(out_format)) { if (!opts->audio_output_channels.set || opts->audio_output_channels.auto_safe) ao_flags |= AO_INIT_SAFE_MULTICHANNEL_ONLY;
mp_chmap_sel_list(&out_channels, opts->audio_output_channels.chmaps, opts->audio_output_channels.num_chmaps); }
mpctx->ao = ao_init_best(mpctx->global, ao_flags, mp_wakeup_core_cb, mpctx, mpctx->encode_lavc_ctx, out_rate, out_format, out_channels); ao_c->ao = mpctx->ao;
int ao_rate = 0; int ao_format = 0; struct mp_chmap ao_channels = {0}; if (mpctx->ao) ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
// Verify passthrough format was not changed. if (mpctx->ao && af_fmt_is_spdif(out_format)) { if (out_rate != ao_rate || out_format != ao_format || !mp_chmap_equals(&out_channels, &ao_channels)) { MP_ERR(mpctx, "Passthrough format unsupported.\n"); ao_uninit(mpctx->ao); mpctx->ao = NULL; ao_c->ao = NULL; } }
if (!mpctx->ao) { // If spdif was used, try to fallback to PCM. if (spdif_fallback && ao_c->audio_src) { 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)) goto init_error; reset_audio_state(mpctx); mp_aframe_reset(ao_c->input_format); mp_wakeup_core(mpctx); // reinit with new format next time return; }
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n"); mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED; goto init_error; }
mp_audio_buffer_reinit_fmt(ao_c->ao_buffer, ao_format, &ao_channels, ao_rate);
#if HAVE_LIBAF afs->output = (struct mp_audio){0}; afs->output.rate = ao_rate; mp_audio_set_format(&afs->output, ao_format); mp_audio_set_channels(&afs->output, &ao_channels); if (!mp_audio_config_equals(&afs->output, &afs->filter_output)) afs->initialized = 0;#else int in_rate = mp_aframe_get_rate(ao_c->input_format); int in_format = mp_aframe_get_format(ao_c->input_format); struct mp_chmap in_chmap = {0}; mp_aframe_get_chmap(ao_c->input_format, &in_chmap); if (!mp_aconverter_reconfig(ao_c->conv, in_rate, in_format, in_chmap, ao_rate, ao_format, ao_channels)) { MP_ERR(mpctx, "Cannot convert audio data for output.\n"); goto init_error; } ao_c->filter_input_format = mp_aframe_new_ref(ao_c->input_format);#endif
mpctx->ao_decoder_fmt = mp_aframe_new_ref(ao_c->input_format);
char tmp[80]; MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao), audio_config_to_str_buf(tmp, sizeof(tmp), ao_rate, ao_format, ao_channels)); MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao)); update_window_title(mpctx, true);
ao_c->ao_resume_time = opts->audio_wait_open > 0 ? mp_time_sec() + opts->audio_wait_open : 0; }
#if HAVE_LIBAF if (recreate_audio_filters(mpctx) < 0) goto init_error;#endif
update_playback_speed(mpctx); audio_update_volume(mpctx);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
return;
init_error: uninit_audio_chain(mpctx); uninit_audio_out(mpctx); error_on_track(mpctx, track);}
int init_audio_decoder(struct MPContext *mpctx, struct track *track){ assert(!track->d_audio); 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;
d_audio->try_spdif = true;
if (!audio_init_best_codec(d_audio)) goto init_error;
return 1;
init_error: if (track->sink) lavfi_set_connected(track->sink, false); track->sink = NULL; audio_uninit(track->d_audio); track->d_audio = NULL; error_on_track(mpctx, track); return 0;}
void reinit_audio_chain(struct MPContext *mpctx){ struct track *track = NULL; track = mpctx->current_track[0][STREAM_AUDIO]; if (!track || !track->stream) { uninit_audio_out(mpctx); error_on_track(mpctx, track); return; } reinit_audio_chain_src(mpctx, track);}
// (track=NULL creates a blank chain, used for lavfi-complex)void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track){ assert(!mpctx->ao_chain);
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
struct ao_chain *ao_c = talloc_zero(NULL, struct ao_chain); mpctx->ao_chain = ao_c; ao_c->log = mpctx->log;#if HAVE_LIBAF ao_c->af = af_new(mpctx->global);#else ao_c->conv = mp_aconverter_create(mpctx->global, mpctx->log, NULL);#endif ao_c->spdif_passthrough = true; ao_c->pts = MP_NOPTS_VALUE; ao_c->ao_buffer = mp_audio_buffer_create(NULL); ao_c->ao = mpctx->ao; ao_c->input_format = mp_aframe_create();
if (track) { ao_c->track = track; track->ao_c = ao_c; if (!init_audio_decoder(mpctx, track)) goto init_error; ao_c->audio_src = track->d_audio; }
reset_audio_state(mpctx);
if (mpctx->ao) { int rate; int format; struct mp_chmap channels; ao_get_format(mpctx->ao, &rate, &format, &channels); mp_audio_buffer_reinit_fmt(ao_c->ao_buffer, format, &channels, rate);
audio_update_volume(mpctx); }
mp_wakeup_core(mpctx); return;
init_error: uninit_audio_chain(mpctx); uninit_audio_out(mpctx); error_on_track(mpctx, track);}
// Return pts value corresponding to the end point of audio written to the// ao so far.double written_audio_pts(struct MPContext *mpctx){ struct ao_chain *ao_c = mpctx->ao_chain; if (!ao_c) return MP_NOPTS_VALUE;
// first calculate the end pts of audio that has been output by decoder double a_pts = ao_c->pts; if (a_pts == MP_NOPTS_VALUE) return MP_NOPTS_VALUE;
// Data buffered in audio filters, measured in seconds of "missing" output double buffered_output = 0;
#if HAVE_LIBAF if (ao_c->af->initialized < 1) return MP_NOPTS_VALUE;
buffered_output += af_calc_delay(ao_c->af);#endif
if (ao_c->conv) buffered_output += mp_aconverter_get_latency(ao_c->conv);
if (ao_c->output_frame) buffered_output += mp_aframe_duration(ao_c->output_frame);
// 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(ao_c->ao_buffer);
// Filters divide audio length by audio_speed, so multiply by it // to get the length in original units without speedup or slowdown a_pts -= buffered_output * mpctx->audio_speed;
return a_pts;}
// 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 || !mpctx->ao) return pts; return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);}
static int write_to_ao(struct MPContext *mpctx, uint8_t **planes, int samples, int flags){ if (mpctx->paused) return 0; struct ao *ao = mpctx->ao; int samplerate; int format; struct mp_chmap channels; ao_get_format(ao, &samplerate, &format, &channels);#if HAVE_ENCODING encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));#endif if (samples == 0) return 0; double real_samplerate = samplerate / mpctx->audio_speed; int played = ao_play(mpctx->ao, (void **)planes, samples, flags); assert(played <= samples); if (played > 0) { mpctx->shown_aframes += played; mpctx->delay += played / real_samplerate; mpctx->written_audio += played / (double)samplerate; return played; } return 0;}
static void dump_audio_stats(struct MPContext *mpctx){ if (!mp_msg_test(mpctx->log, MSGL_STATS)) return; if (mpctx->audio_status != STATUS_PLAYING || !mpctx->ao || mpctx->paused) { mpctx->audio_stat_start = 0; return; }
double delay = ao_get_delay(mpctx->ao); if (!mpctx->audio_stat_start) { mpctx->audio_stat_start = mp_time_us(); mpctx->written_audio = delay; } double current_audio = mpctx->written_audio - delay; double current_time = (mp_time_us() - mpctx->audio_stat_start) / 1e6; MP_STATS(mpctx, "value %f ao-dev", current_audio - current_time);}
// Return the number of samples that must be skipped or prepended to reach the// target audio pts after a seek (for A/V sync or hr-seek).// Return value (*skip):// >0: skip this many samples// =0: don't do anything// <0: prepend this many samples of silence// Returns false if PTS is not known yet.static bool get_sync_samples(struct MPContext *mpctx, int *skip){ struct MPOpts *opts = mpctx->opts; *skip = 0;
if (mpctx->audio_status != STATUS_SYNCING) return true;
int ao_rate; int ao_format; struct mp_chmap ao_channels; ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);
double play_samplerate = ao_rate / mpctx->audio_speed;
if (!opts->initial_audio_sync) { mpctx->audio_status = STATUS_FILLING; return true; }
double written_pts = written_audio_pts(mpctx); if (written_pts == MP_NOPTS_VALUE && !mp_audio_buffer_samples(mpctx->ao_chain->ao_buffer)) return false; // no audio read yet
bool sync_to_video = mpctx->vo_chain && !mpctx->vo_chain->is_coverart && mpctx->video_status != STATUS_EOF;
double sync_pts = MP_NOPTS_VALUE; if (sync_to_video) { if (mpctx->video_status < STATUS_READY) return false; // wait until we know a video PTS if (mpctx->video_pts != MP_NOPTS_VALUE) sync_pts = mpctx->video_pts - opts->audio_delay; } else if (mpctx->hrseek_active) { sync_pts = mpctx->hrseek_pts; } else { // If audio-only is enabled mid-stream during playback, sync accordingly. sync_pts = mpctx->playback_pts; } if (sync_pts == MP_NOPTS_VALUE) { mpctx->audio_status = STATUS_FILLING; return true; // syncing disabled }
double ptsdiff = written_pts - sync_pts; // Missing timestamp, or PTS reset, or just broken. if (written_pts == MP_NOPTS_VALUE) { MP_WARN(mpctx, "Failed audio resync.\n"); mpctx->audio_status = STATUS_FILLING; return true; } ptsdiff = MPCLAMP(ptsdiff, -3600, 3600);
// Heuristic: if audio is "too far" ahead, and one of them is a separate // track, allow a refresh seek to the correct position to fix it. if (ptsdiff > 0.2 && mpctx->audio_allow_second_chance_seek && sync_to_video) { struct ao_chain *ao_c = mpctx->ao_chain; if (ao_c && ao_c->track && mpctx->vo_chain && mpctx->vo_chain->track && ao_c->track->demuxer != mpctx->vo_chain->track->demuxer) { struct track *track = ao_c->track; double pts = mpctx->video_pts; if (pts != MP_NOPTS_VALUE) pts += get_track_seek_offset(mpctx, track); // (disable it first to make it take any effect) demuxer_select_track(track->demuxer, track->stream, pts, false); demuxer_select_track(track->demuxer, track->stream, pts, true); reset_audio_state(mpctx); MP_VERBOSE(mpctx, "retrying audio seek\n"); return false; } } mpctx->audio_allow_second_chance_seek = false;
int align = af_format_sample_alignment(ao_format); *skip = (int)(-ptsdiff * play_samplerate) / align * align; return true;}
static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c, int minsamples, double endpts, bool eof, bool *seteof){ struct mp_audio_buffer *outbuf = ao_c->ao_buffer;
int ao_rate; int ao_format; struct mp_chmap ao_channels; ao_get_format(ao_c->ao, &ao_rate, &ao_format, &ao_channels);
while (mp_audio_buffer_samples(outbuf) < minsamples) { int cursamples = mp_audio_buffer_samples(outbuf); int maxsamples = INT_MAX; if (endpts != MP_NOPTS_VALUE) { double rate = ao_rate / mpctx->audio_speed; double curpts = written_audio_pts(mpctx); if (curpts != MP_NOPTS_VALUE) { double remaining = (endpts - curpts - mpctx->opts->audio_delay) * rate; maxsamples = MPCLAMP(remaining, 0, INT_MAX); } }
if (!ao_c->output_frame || !mp_aframe_get_size(ao_c->output_frame)) { TA_FREEP(&ao_c->output_frame);#if HAVE_LIBAF struct af_stream *afs = mpctx->ao_chain->af; if (af_output_frame(afs, eof) < 0) return true; // error, stop doing stuff struct mp_audio *mpa = af_read_output_frame(afs); ao_c->output_frame = mp_audio_to_aframe(mpa); talloc_free(mpa);#else if (eof) mp_aconverter_write_input(ao_c->conv, NULL); mp_aconverter_set_speed(ao_c->conv, mpctx->audio_speed); bool got_eof; ao_c->output_frame = mp_aconverter_read_output(ao_c->conv, &got_eof);#endif }
if (!ao_c->output_frame) return false; // out of data
if (cursamples + mp_aframe_get_size(ao_c->output_frame) > maxsamples) { if (cursamples < maxsamples) { uint8_t **data = mp_aframe_get_data_ro(ao_c->output_frame); mp_audio_buffer_append(outbuf, (void **)data, maxsamples - cursamples); mp_aframe_skip_samples(ao_c->output_frame, maxsamples - cursamples); } *seteof = true; return true; }
uint8_t **data = mp_aframe_get_data_ro(ao_c->output_frame); mp_audio_buffer_append(outbuf, (void **)data, mp_aframe_get_size(ao_c->output_frame)); TA_FREEP(&ao_c->output_frame); } 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) { res = lavfi_request_frame_a(ao_c->filter_src, &ao_c->input_frame); } else if (ao_c->audio_src) { audio_work(ao_c->audio_src); res = audio_get_frame(ao_c->audio_src, &ao_c->input_frame); }
if (ao_c->input_frame) mp_aframe_config_copy(ao_c->input_format, 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. * In the former case outbuf has at least minsamples buffered on return. * In case of EOF/error it might or might not be. */static int filter_audio(struct MPContext *mpctx, struct mp_audio_buffer *outbuf, int minsamples){ struct ao_chain *ao_c = mpctx->ao_chain;#if HAVE_LIBAF struct af_stream *afs = ao_c->af; if (afs->initialized < 1) return AD_ERR;#else if (!ao_c->filter_input_format) return AD_ERR;#endif
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) copy_output(mpctx, ao_c, minsamples, endpts, true, &eof); break; }
// On format change, make sure to drain the filter chain.#if HAVE_LIBAF struct mp_audio in_format; mp_audio_config_from_aframe(&in_format, ao_c->input_format); if (!mp_audio_config_equals(&afs->input, &in_format)) { copy_output(mpctx, ao_c, minsamples, endpts, true, &eof); res = AD_NEW_FMT; break; }#else if (!mp_aframe_config_equals(ao_c->filter_input_format, ao_c->input_format)) { copy_output(mpctx, ao_c, minsamples, endpts, true, &eof); res = AD_NEW_FMT; break; }#endif
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 HAVE_LIBAF struct mp_audio *mpa = mp_audio_from_aframe(ao_c->input_frame); talloc_free(ao_c->input_frame); ao_c->input_frame = NULL; if (!mpa) abort(); if (af_filter_frame(afs, mpa) < 0) return AD_ERR;#else if (mp_aconverter_write_input(ao_c->conv, ao_c->input_frame)) ao_c->input_frame = NULL;#endif }
if (res == 0 && mp_audio_buffer_samples(outbuf) < minsamples && eof) res = AD_EOF;
MP_STATS(ao_c, "end audio");
return res;}
void reload_audio_output(struct MPContext *mpctx){ if (!mpctx->ao) return;
ao_reset(mpctx->ao); uninit_audio_out(mpctx); reinit_audio_filters(mpctx); // mostly to issue refresh seek
// 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) { ao_c->spdif_passthrough = true; ao_c->spdif_failed = false; d_audio->try_spdif = true;#if HAVE_LIBAF ao_c->af->initialized = 0;#endif TA_FREEP(&ao_c->filter_input_format); if (!audio_init_best_codec(d_audio)) { MP_ERR(mpctx, "Error reinitializing audio.\n"); error_on_track(mpctx, ao_c->track); } } }
mp_wakeup_core(mpctx);}
void fill_audio_out_buffers(struct MPContext *mpctx){ struct MPOpts *opts = mpctx->opts; bool was_eof = mpctx->audio_status == STATUS_EOF;
dump_audio_stats(mpctx);
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) reload_audio_output(mpctx);
struct ao_chain *ao_c = mpctx->ao_chain; if (!ao_c) return;
bool is_initialized = !!ao_c->filter_input_format;#if HAVE_LIBAF is_initialized = ao_c->af->initialized == 1;#endif
if (!is_initialized || !mpctx->ao) { // 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; } reinit_audio_filters_and_output(mpctx); mp_wakeup_core(mpctx); return; // try again next iteration }
if (ao_c->ao_resume_time > mp_time_sec()) { double remaining = ao_c->ao_resume_time - mp_time_sec(); mp_set_timeout(mpctx, remaining); return; }
if (mpctx->vo_chain && ao_c->pts_reset) { MP_VERBOSE(mpctx, "Reset playback due to audio timestamp reset.\n"); reset_playback_state(mpctx); mp_wakeup_core(mpctx); return; }
int ao_rate; int ao_format; struct mp_chmap ao_channels; ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels); double play_samplerate = ao_rate / mpctx->audio_speed; int align = af_format_sample_alignment(ao_format);
// If audio is infinitely fast, somehow try keeping approximate A/V sync. if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) && mpctx->video_status != STATUS_EOF && mpctx->delay > 0) return;
int playsize = ao_get_space(mpctx->ao);
int skip = 0; bool sync_known = get_sync_samples(mpctx, &skip); if (skip > 0) { playsize = MPMIN(skip + 1, MPMAX(playsize, 2500)); // buffer extra data } else if (skip < 0) { playsize = MPMAX(1, playsize + skip); // silence will be prepended }
int skip_duplicate = 0; // >0: skip, <0: duplicate double drop_limit = (opts->sync_max_audio_change + opts->sync_max_video_change) / 100; if (mpctx->display_sync_active && opts->video_sync == VS_DISP_ADROP && fabs(mpctx->last_av_difference) >= opts->sync_audio_drop_size && mpctx->audio_drop_throttle < drop_limit && mpctx->audio_status == STATUS_PLAYING) { int samples = ceil(opts->sync_audio_drop_size * play_samplerate); samples = (samples + align / 2) / align * align;
skip_duplicate = mpctx->last_av_difference >= 0 ? -samples : samples;
playsize = MPMAX(playsize, samples);
mpctx->audio_drop_throttle += 1 - drop_limit - samples / play_samplerate; }
playsize = playsize / align * align;
int status = mpctx->audio_status >= STATUS_DRAINING ? AD_EOF : AD_OK; bool working = false; if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) { status = filter_audio(mpctx, ao_c->ao_buffer, playsize); if (status == AD_WAIT) return; if (status == AD_NO_PROGRESS || status == AD_STARVE) { mp_wakeup_core(mpctx); return; } if (status == AD_NEW_FMT) { /* 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 < 1) uninit_audio_out(mpctx); reinit_audio_filters_and_output(mpctx); mp_wakeup_core(mpctx); return; // retry on next iteration } if (status == AD_ERR) mp_wakeup_core(mpctx); working = true; }
// If EOF was reached before, but now something can be decoded, try to // restart audio properly. This helps with video files where audio starts // later. Retrying is needed to get the correct sync PTS. if (mpctx->audio_status >= STATUS_DRAINING && mp_audio_buffer_samples(ao_c->ao_buffer) > 0) { mpctx->audio_status = STATUS_SYNCING; return; // retry on next iteration }
bool end_sync = false; if (skip >= 0) { int max = mp_audio_buffer_samples(ao_c->ao_buffer); mp_audio_buffer_skip(ao_c->ao_buffer, MPMIN(skip, max)); // If something is left, we definitely reached the target time. end_sync |= sync_known && skip < max; working |= skip > 0; } else if (skip < 0) { if (-skip > playsize) { // heuristic against making the buffer too large ao_reset(mpctx->ao); // some AOs repeat data on underflow mpctx->audio_status = STATUS_DRAINING; mpctx->delay = 0; return; } mp_audio_buffer_prepend_silence(ao_c->ao_buffer, -skip); end_sync = true; }
if (skip_duplicate) { int max = mp_audio_buffer_samples(ao_c->ao_buffer); if (abs(skip_duplicate) > max) skip_duplicate = skip_duplicate >= 0 ? max : -max; mpctx->last_av_difference += skip_duplicate / play_samplerate; if (skip_duplicate >= 0) { mp_audio_buffer_skip(ao_c->ao_buffer, skip_duplicate); MP_STATS(mpctx, "drop-audio"); } else { mp_audio_buffer_duplicate(ao_c->ao_buffer, -skip_duplicate); MP_STATS(mpctx, "duplicate-audio"); } MP_VERBOSE(mpctx, "audio skip_duplicate=%d\n", skip_duplicate); }
if (mpctx->audio_status == STATUS_SYNCING) { if (end_sync) mpctx->audio_status = STATUS_FILLING; if (status != AD_OK && !mp_audio_buffer_samples(ao_c->ao_buffer)) mpctx->audio_status = STATUS_EOF; if (working || end_sync) mp_wakeup_core(mpctx); return; // continue on next iteration }
assert(mpctx->audio_status >= STATUS_FILLING);
// We already have as much data as the audio device wants, and can start // writing it any time. if (mpctx->audio_status == STATUS_FILLING) mpctx->audio_status = STATUS_READY;
// Even if we're done decoding and syncing, let video start first - this is // required, because sending audio to the AO already starts playback. if (mpctx->audio_status == STATUS_READY) { if (mpctx->vo_chain && !mpctx->vo_chain->is_coverart && mpctx->video_status <= STATUS_READY) return; MP_VERBOSE(mpctx, "starting audio playback\n"); }
bool audio_eof = status == AD_EOF; bool partial_fill = false; int playflags = 0;
if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) { playsize = mp_audio_buffer_samples(ao_c->ao_buffer); partial_fill = true; }
audio_eof &= partial_fill;
// With gapless audio, delay this to ao_uninit. There must be only // 1 final chunk, and that is handled when calling ao_uninit(). if (audio_eof && !opts->gapless_audio) playflags |= AOPLAY_FINAL_CHUNK;
uint8_t **planes; int samples; mp_audio_buffer_peek(ao_c->ao_buffer, &planes, &samples); if (audio_eof || samples >= align) samples = samples / align * align; samples = MPMIN(samples, mpctx->paused ? 0 : playsize); int played = write_to_ao(mpctx, planes, samples, playflags); assert(played >= 0 && played <= samples); mp_audio_buffer_skip(ao_c->ao_buffer, played);
mpctx->audio_drop_throttle = MPMAX(0, mpctx->audio_drop_throttle - played / play_samplerate);
dump_audio_stats(mpctx);
mpctx->audio_status = STATUS_PLAYING; if (audio_eof && !playsize) { mpctx->audio_status = STATUS_DRAINING; // Wait until the AO has played all queued data. In the gapless case, // we trigger EOF immediately, and let it play asynchronously. if (ao_eof_reached(mpctx->ao) || opts->gapless_audio) { mpctx->audio_status = STATUS_EOF; if (!was_eof) { MP_VERBOSE(mpctx, "audio EOF reached\n"); mp_wakeup_core(mpctx); } } }}
// 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);}