From e44911142914783c9ec717f329bd9b6a8bb9b70e Mon Sep 17 00:00:00 2001 From: wm4 Date: Tue, 17 Dec 2013 00:53:22 +0100 Subject: Move mpvcore/player/ to player/ --- player/audio.c | 471 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 471 insertions(+) create mode 100644 player/audio.c (limited to 'player/audio.c') 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 +#include +#include +#include +#include + +#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); +} -- cgit v1.2.3