diff options
Diffstat (limited to 'player/audio.c')
| -rw-r--r-- | player/audio.c | 98 |
1 files changed, 56 insertions, 42 deletions
diff --git a/player/audio.c b/player/audio.c index 984243b639..c6e7411e2a 100644 --- a/player/audio.c +++ b/player/audio.c @@ -45,7 +45,6 @@ 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) @@ -54,13 +53,16 @@ static int build_afilter_chain(struct MPContext *mpctx) struct mp_audio in_format; mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format); + struct mp_audio out_format; + ao_get_format(mpctx->ao, &out_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) { + if (new_srate != out_format.rate) { // limits are taken from libaf/af_resample.c if (new_srate < 8000) new_srate = 8000; @@ -70,7 +72,7 @@ static int build_afilter_chain(struct MPContext *mpctx) } } return audio_init_filters(d_audio, new_srate, - &ao->samplerate, &ao->channels, &ao->format); + &out_format.rate, &out_format.channels, &out_format.format); } static int recreate_audio_filters(struct MPContext *mpctx) @@ -135,9 +137,11 @@ void reinit_audio_chain(struct MPContext *mpctx) 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; + struct mp_audio out_format; + ao_get_format(mpctx->ao, &out_format); + ao_srate = out_format.rate; + ao_format = out_format.format; + ao_channels = out_format.channels; } else { // Automatic downmix if (mp_chmap_is_stereo(&opts->audio_output_channels) && @@ -172,14 +176,16 @@ void reinit_audio_chain(struct MPContext *mpctx) goto init_error; } - ao->buffer = mp_audio_buffer_create(ao); - mp_audio_buffer_reinit_fmt(ao->buffer, ao->format, &ao->channels, - ao->samplerate); + struct mp_audio fmt; + ao_get_format(ao, &fmt); - char *s = mp_audio_fmt_to_str(ao->samplerate, &ao->channels, ao->format); - MP_INFO(mpctx, "AO: [%s] %s\n", ao->driver->name, s); + mpctx->ao_buffer = mp_audio_buffer_create(ao); + mp_audio_buffer_reinit(mpctx->ao_buffer, &fmt); + + char *s = mp_audio_config_to_str(&fmt); + MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(ao), s); talloc_free(s); - MP_VERBOSE(mpctx, "AO: Description: %s\n", ao->driver->description); + MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(ao)); update_window_title(mpctx, true); } @@ -228,7 +234,7 @@ double written_audio_pts(struct MPContext *mpctx) // 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); + 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 @@ -252,11 +258,13 @@ static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags if (mpctx->paused) return 0; struct ao *ao = mpctx->ao; - ao->pts = pts; + struct mp_audio out_format; + ao_get_format(ao, &out_format); + mpctx->ao_pts = pts; #if HAVE_ENCODING - encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, ao->pts); + encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, mpctx->ao_pts); #endif - double real_samplerate = ao->samplerate / mpctx->opts->playback_speed; + double real_samplerate = out_format.rate / mpctx->opts->playback_speed; int played = ao_play(mpctx->ao, data->planes, data->samples, flags); assert(played <= data->samples); if (played > 0) { @@ -264,7 +272,7 @@ static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags 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; + mpctx->ao_pts += played / real_samplerate; return played; } return 0; @@ -274,7 +282,7 @@ 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); + 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++) @@ -295,15 +303,18 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize) assert(d_audio); + struct mp_audio out_format; + ao_get_format(ao, &out_format); + // Timing info may not be set without - res = audio_decode(d_audio, ao->buffer, 1); + res = audio_decode(d_audio, mpctx->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; + double real_samplerate = out_format.rate / opts->playback_speed; bool hrseek = mpctx->hrseek_active; // audio only hrseek mpctx->hrseek_active = false; while (1) { @@ -320,7 +331,7 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize) 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); + res = audio_decode(d_audio, mpctx->ao_buffer, out_format.rate); if (res < 0) return res; did_retry = true; @@ -338,16 +349,16 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize) 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); - ao->buffer_playable_samples = 0; + res = audio_decode(d_audio, mpctx->ao_buffer, a); + if (skip_samples <= mp_audio_buffer_samples(mpctx->ao_buffer)) { + mp_audio_buffer_skip(mpctx->ao_buffer, skip_samples); + mpctx->ao_buffer_playable_samples = 0; if (res < 0) return res; - return audio_decode(d_audio, ao->buffer, playsize); + return audio_decode(d_audio, mpctx->ao_buffer, playsize); } - mp_audio_buffer_clear(ao->buffer); - ao->buffer_playable_samples = 0; + mp_audio_buffer_clear(mpctx->ao_buffer); + mpctx->ao_buffer_playable_samples = 0; if (res < 0) return res; } @@ -357,15 +368,15 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize) 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 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); + mp_audio_buffer_prepend_silence(mpctx->ao_buffer, samples); + return audio_decode(d_audio, mpctx->ao_buffer, playsize); } int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) @@ -378,8 +389,10 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) bool signal_eof = false; bool partial_fill = false; struct dec_audio *d_audio = mpctx->d_audio; + struct mp_audio out_format; + ao_get_format(ao, &out_format); // Can't adjust the start of audio with spdif pass-through. - bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK); + bool modifiable_audio_format = !(out_format.format & AF_FORMAT_SPECIAL_MASK); assert(d_audio); @@ -400,7 +413,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) if (mpctx->syncing_audio || mpctx->hrseek_active) res = audio_start_sync(mpctx, playsize); else - res = audio_decode(d_audio, ao->buffer, playsize); + res = audio_decode(d_audio, mpctx->ao_buffer, playsize); if (res < 0) { // EOF, error or format change if (res == -2) { @@ -420,7 +433,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) if (endpts != MP_NOPTS_VALUE) { double samples = (endpts - written_audio_pts(mpctx) - mpctx->audio_delay) - * ao->samplerate / opts->playback_speed; + * out_format.rate / opts->playback_speed; if (playsize > samples) { playsize = MPMAX(samples, 0); audio_eof = true; @@ -428,8 +441,8 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) } } - if (playsize > mp_audio_buffer_samples(ao->buffer)) { - playsize = mp_audio_buffer_samples(ao->buffer); + if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) { + playsize = mp_audio_buffer_samples(mpctx->ao_buffer); partial_fill = true; } if (!playsize) @@ -445,17 +458,18 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) } } - assert(ao->buffer_playable_samples <= mp_audio_buffer_samples(ao->buffer)); + assert(mpctx->ao_buffer_playable_samples <= + mp_audio_buffer_samples(mpctx->ao_buffer)); struct mp_audio data; - mp_audio_buffer_peek(ao->buffer, &data); + mp_audio_buffer_peek(mpctx->ao_buffer, &data); data.samples = MPMIN(data.samples, playsize); int played = write_to_ao(mpctx, &data, playflags, written_audio_pts(mpctx)); assert(played >= 0 && played <= data.samples); - ao->buffer_playable_samples = playsize - played; + mpctx->ao_buffer_playable_samples = playsize - played; if (played > 0) { - mp_audio_buffer_skip(ao->buffer, played); + mp_audio_buffer_skip(mpctx->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 @@ -470,8 +484,8 @@ 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; + mp_audio_buffer_clear(mpctx->ao_buffer); + mpctx->ao_buffer_playable_samples = 0; } } |