diff options
Diffstat (limited to 'mpvcore/player/audio.c')
-rw-r--r-- | mpvcore/player/audio.c | 414 |
1 files changed, 414 insertions, 0 deletions
diff --git a/mpvcore/player/audio.c b/mpvcore/player/audio.c new file mode 100644 index 0000000000..443a1eed90 --- /dev/null +++ b/mpvcore/player/audio.c @@ -0,0 +1,414 @@ +/* + * 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/decode/dec_audio.h" +#include "audio/filter/af.h" +#include "audio/out/ao.h" +#include "demux/demux.h" + +#include "mp_core.h" + +static int build_afilter_chain(struct MPContext *mpctx) +{ + struct sh_audio *sh_audio = mpctx->sh_audio; + struct ao *ao = mpctx->ao; + struct MPOpts *opts = mpctx->opts; + int new_srate; + if (af_control_any_rev(sh_audio->afilter, + AF_CONTROL_PLAYBACK_SPEED | AF_CONTROL_SET, + &opts->playback_speed)) + new_srate = sh_audio->samplerate; + else { + new_srate = sh_audio->samplerate * 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 = (double)new_srate / sh_audio->samplerate; + } + } + return init_audio_filters(sh_audio, new_srate, + &ao->samplerate, &ao->channels, &ao->format); +} + +static int recreate_audio_filters(struct MPContext *mpctx) +{ + assert(mpctx->sh_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->sh_audio->afilter); + + return 0; +} + +int reinit_audio_filters(struct MPContext *mpctx) +{ + struct sh_audio *sh_audio = mpctx->sh_audio; + if (!sh_audio) + return -2; + + af_uninit(mpctx->sh_audio->afilter); + if (af_init(mpctx->sh_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; + init_demux_stream(mpctx, STREAM_AUDIO); + if (!mpctx->sh_audio) { + uninit_player(mpctx, INITIALIZED_AO); + goto no_audio; + } + + if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) { + if (!init_best_audio_codec(mpctx->sh_audio, opts->audio_decoders)) + goto init_error; + mpctx->initialized_flags |= INITIALIZED_ACODEC; + } + + 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(&mpctx->sh_audio->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 (!init_audio_filters(mpctx->sh_audio, // preliminary init + // input: + mpctx->sh_audio->samplerate, + // 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.start = talloc_new(ao); + 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); + } + + 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) +{ + sh_audio_t *sh_audio = mpctx->sh_audio; + if (!sh_audio) + return MP_NOPTS_VALUE; + + double bps = sh_audio->channels.num * sh_audio->samplerate * + sh_audio->samplesize; + + // first calculate the end pts of audio that has been output by decoder + double a_pts = sh_audio->pts; + if (a_pts == MP_NOPTS_VALUE) + return MP_NOPTS_VALUE; + + // sh_audio->pts is the timestamp of the latest input packet with + // known pts that the decoder has decoded. sh_audio->pts_bytes is + // the amount of bytes the decoder has written after that timestamp. + a_pts += sh_audio->pts_bytes / bps; + + // 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 -= sh_audio->a_buffer_len / bps; + + // Data buffered in audio filters, measured in bytes of "missing" output + double buffered_output = af_calc_delay(sh_audio->afilter); + + // Data that was ready for ao but was buffered because ao didn't fully + // accept everything to internal buffers yet + buffered_output += mpctx->ao->buffer.len; + + // 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 / mpctx->ao->bps; + + 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, void *data, int len, int flags, + double pts) +{ + if (mpctx->paused) + return 0; + struct ao *ao = mpctx->ao; + double bps = ao->bps / mpctx->opts->playback_speed; + ao->pts = pts; + int played = ao_play(mpctx->ao, data, len, flags); + if (played > 0) { + mpctx->shown_aframes += played / (af_fmt2bits(ao->format) / 8); + mpctx->delay += played / bps; + // Keep correct pts for remaining data - could be used to flush + // remaining buffer when closing ao. + ao->pts += played / bps; + return played; + } + return 0; +} + +#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; + sh_audio_t * const sh_audio = mpctx->sh_audio; + int res; + + // Timing info may not be set without + res = decode_audio(sh_audio, &ao->buffer, 1); + if (res < 0) + return res; + + int bytes; + bool did_retry = false; + double written_pts; + double bps = ao->bps / 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->sh_video->pts - mpctx->delay + - mpctx->audio_delay; + bytes = ptsdiff * bps; + bytes -= bytes % (ao->channels.num * af_fmt2bits(ao->format) / 8); + + // ogg demuxers give packets without timing + if (written_pts <= 1 && sh_audio->pts == MP_NOPTS_VALUE) { + if (!did_retry) { + // Try to read more data to see packets that have pts + res = decode_audio(sh_audio, &ao->buffer, ao->bps); + if (res < 0) + return res; + did_retry = true; + continue; + } + bytes = 0; + } + + if (fabs(ptsdiff) > 300 || isnan(ptsdiff)) // pts reset or just broken? + bytes = 0; + + if (bytes > 0) + break; + + mpctx->syncing_audio = false; + int a = MPMIN(-bytes, MPMAX(playsize, 20000)); + res = decode_audio(sh_audio, &ao->buffer, a); + bytes += ao->buffer.len; + if (bytes >= 0) { + memmove(ao->buffer.start, + ao->buffer.start + ao->buffer.len - bytes, bytes); + ao->buffer.len = bytes; + if (res < 0) + return res; + return decode_audio(sh_audio, &ao->buffer, playsize); + } + ao->buffer.len = 0; + if (res < 0) + return res; + } + if (hrseek) + // Don't add silence in audio-only case even if position is too late + return 0; + int fillbyte = 0; + if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_US) + fillbyte = 0x80; + if (bytes >= playsize) { + /* This case could fall back to the one below with + * bytes = playsize, but then silence would keep accumulating + * in a_out_buffer if the AO accepts less data than it asks for + * in playsize. */ + char *p = malloc(playsize); + memset(p, fillbyte, playsize); + write_to_ao(mpctx, p, playsize, 0, written_pts - bytes / bps); + free(p); + return ASYNC_PLAY_DONE; + } + mpctx->syncing_audio = false; + decode_audio_prepend_bytes(&ao->buffer, bytes, fillbyte); + return decode_audio(sh_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 partial_fill = false; + sh_audio_t * const sh_audio = mpctx->sh_audio; + bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK); + int unitsize = ao->channels.num * af_fmt2bits(ao->format) / 8; + + 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->sh_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 = decode_audio(sh_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(mpctx->sh_audio->gsh)) + audio_eof = true; + } + + if (endpts != MP_NOPTS_VALUE && modifiable_audio_format) { + double bytes = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay) + * ao->bps / opts->playback_speed; + if (playsize > bytes) { + playsize = MPMAX(bytes, 0); + playflags |= AOPLAY_FINAL_CHUNK; + audio_eof = true; + partial_fill = true; + } + } + + assert(ao->buffer.len % unitsize == 0); + if (playsize > ao->buffer.len) { + partial_fill = true; + playsize = ao->buffer.len; + if (audio_eof) + playflags |= AOPLAY_FINAL_CHUNK; + } + playsize -= playsize % unitsize; + if (!playsize) + return partial_fill && audio_eof ? -2 : -partial_fill; + + // play audio: + + int played = write_to_ao(mpctx, ao->buffer.start, playsize, playflags, + written_audio_pts(mpctx)); + assert(played % unitsize == 0); + ao->buffer_playable_size = playsize - played; + + if (played > 0) { + ao->buffer.len -= played; + memmove(ao->buffer.start, ao->buffer.start + played, ao->buffer.len); + } 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 + return -2; + } + + return -partial_fill; +} |