From 3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Thu, 21 Sep 2017 11:07:02 +0200 Subject: audio: move libswresample wrapper out of audio filter code Move it from af_lavrresample.c to a new aconverter.c file, which is independent from the filter chain code. It also doesn't use mp_audio, and thus has no GPL dependencies. Preparation for later commits. Not particularly well tested, so have fun. --- audio/filter/af_lavrresample.c | 492 +++-------------------------------------- 1 file changed, 34 insertions(+), 458 deletions(-) (limited to 'audio/filter/af_lavrresample.c') diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index c18f8cc28a..55eb6b0f20 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -27,326 +27,24 @@ #include #include -#include -#include -#include -#include -#include - #include "common/common.h" #include "config.h" -#define HAVE_LIBSWRESAMPLE HAVE_IS_FFMPEG -#define HAVE_LIBAVRESAMPLE HAVE_IS_LIBAV - -#if HAVE_LIBAVRESAMPLE -#include -#elif HAVE_LIBSWRESAMPLE -#include -#define AVAudioResampleContext SwrContext -#define avresample_alloc_context swr_alloc -#define avresample_open swr_init -#define avresample_close(x) do { } while(0) -#define avresample_free swr_free -#define avresample_available(x) 0 -#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \ - swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples) -#define avresample_set_channel_mapping swr_set_channel_mapping -#define avresample_set_compensation swr_set_compensation -#else -#error "config.h broken or no resampler found" -#endif - #include "common/av_common.h" #include "common/msg.h" #include "options/m_option.h" #include "audio/filter/af.h" #include "audio/fmt-conversion.h" #include "osdep/endian.h" - -struct af_resample_opts { - int filter_size; - int phase_shift; - int linear; - double cutoff; - int normalize; -}; +#include "audio/aconverter.h" struct af_resample { int allow_detach; - char **avopts; double playback_speed; - bool is_resampling; - struct AVAudioResampleContext *avrctx; - struct mp_audio avrctx_fmt; // output format of avrctx - struct mp_audio pool_fmt; // format used to allocate frames for avrctx output - struct mp_audio pre_out_fmt; // format before final conversion (S24) - struct AVAudioResampleContext *avrctx_out; // for output channel reordering - struct af_resample_opts opts; // opts requested by the user - // At least libswresample keeps a pointer around for this: - int reorder_in[MP_NUM_CHANNELS]; - int reorder_out[MP_NUM_CHANNELS]; - struct mp_audio_pool *reorder_buffer; - - int in_rate_af; // filter input sample rate - int in_rate; // actual rate (used by lavr), adjusted for playback speed - int in_format; - struct mp_chmap in_channels; - int out_rate; - int out_format; - struct mp_chmap out_channels; -}; - -#if HAVE_LIBAVRESAMPLE -static double get_delay(struct af_resample *s) -{ - return avresample_get_delay(s->avrctx) / (double)s->in_rate + - avresample_available(s->avrctx) / (double)s->out_rate; -} -static int get_out_samples(struct af_resample *s, int in_samples) -{ - return avresample_get_out_samples(s->avrctx, in_samples); -} -#else -static double get_delay(struct af_resample *s) -{ - int64_t base = s->in_rate * (int64_t)s->out_rate; - return swr_get_delay(s->avrctx, base) / (double)base; -} -static int get_out_samples(struct af_resample *s, int in_samples) -{ - return swr_get_out_samples(s->avrctx, in_samples); -} -#endif - -static void close_lavrr(struct af_instance *af) -{ - struct af_resample *s = af->priv; - - if (s->avrctx) - avresample_close(s->avrctx); - avresample_free(&s->avrctx); - if (s->avrctx_out) - avresample_close(s->avrctx_out); - avresample_free(&s->avrctx_out); -} - -static int resample_frame(struct AVAudioResampleContext *r, - struct mp_audio *out, struct mp_audio *in) -{ - return avresample_convert(r, - out ? (uint8_t **)out->planes : NULL, - out ? mp_audio_get_allocated_size(out) : 0, - out ? out->samples : 0, - in ? (uint8_t **)in->planes : NULL, - in ? mp_audio_get_allocated_size(in) : 0, - in ? in->samples : 0); -} - -static double af_resample_default_cutoff(int filter_size) -{ - return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80); -} - -static int rate_from_speed(int rate, double speed) -{ - return lrint(rate * speed); -} - -static struct mp_chmap fudge_pairs[][2] = { - {MP_CHMAP2(BL, BR), MP_CHMAP2(SL, SR)}, - {MP_CHMAP2(SL, SR), MP_CHMAP2(BL, BR)}, - {MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL, SR)}, - {MP_CHMAP2(SL, SR), MP_CHMAP2(SDL, SDR)}, + struct mp_resample_opts opts; + struct mp_aconverter *converter; }; -// Modify out_layout and return the new value. The intention is reducing the -// loss libswresample's rematrixing will cause by exchanging similar, but -// strictly speaking incompatible channel pairs. For example, 7.1 should be -// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave -// it to libswresample to create the remix matrix.) -static uint64_t fudge_layout_conversion(struct af_instance *af, - uint64_t in, uint64_t out) -{ - for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) { - uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]); - uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]); - if ((in & a) == a && (in & b) == 0 && - (out & a) == 0 && (out & b) == b) - { - out = (out & ~b) | a; - - MP_VERBOSE(af, "Fudge: %s -> %s\n", - mp_chmap_to_str(&fudge_pairs[n][0]), - mp_chmap_to_str(&fudge_pairs[n][1])); - } - } - return out; -} - -// mp_chmap_get_reorder() performs: -// to->speaker[n] = from->speaker[src[n]] -// but libavresample does: -// to->speaker[dst[n]] = from->speaker[n] -static void transpose_order(int *map, int num) -{ - int nmap[MP_NUM_CHANNELS] = {0}; - for (int n = 0; n < num; n++) { - for (int i = 0; i < num; i++) { - if (map[n] == i) - nmap[i] = n; - } - } - memcpy(map, nmap, sizeof(nmap)); -} - -static int configure_lavrr(struct af_instance *af, struct mp_audio *in, - struct mp_audio *out, bool verbose) -{ - struct af_resample *s = af->priv; - - close_lavrr(af); - - s->avrctx = avresample_alloc_context(); - s->avrctx_out = avresample_alloc_context(); - if (!s->avrctx || !s->avrctx_out) - goto error; - - enum AVSampleFormat in_samplefmt = af_to_avformat(in->format); - enum AVSampleFormat out_samplefmt = af_to_avformat(out->format); - enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt); - - if (in_samplefmt == AV_SAMPLE_FMT_NONE || - out_samplefmt == AV_SAMPLE_FMT_NONE || - out_samplefmtp == AV_SAMPLE_FMT_NONE) - goto error; - - s->out_rate = out->rate; - s->in_rate_af = in->rate; - s->in_rate = rate_from_speed(in->rate, s->playback_speed); - s->out_format = out->format; - s->in_format = in->format; - s->out_channels= out->channels; - s->in_channels = in->channels; - - av_opt_set_int(s->avrctx, "filter_size", s->opts.filter_size, 0); - av_opt_set_int(s->avrctx, "phase_shift", s->opts.phase_shift, 0); - av_opt_set_int(s->avrctx, "linear_interp", s->opts.linear, 0); - - av_opt_set_double(s->avrctx, "cutoff", s->opts.cutoff, 0); - - int normalize = s->opts.normalize; - if (normalize < 0) - normalize = af->opts->audio_normalize; -#if HAVE_LIBSWRESAMPLE - av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0); -#else - av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0); -#endif - - if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0) - goto error; - - struct mp_chmap map_in = in->channels; - struct mp_chmap map_out = out->channels; - - // Try not to do any remixing if at least one is "unknown". - if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out)) { - mp_chmap_set_unknown(&map_in, map_in.num); - mp_chmap_set_unknown(&map_out, map_out.num); - } - - // unchecked: don't take any channel reordering into account - uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in); - uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out); - - struct mp_chmap in_lavc, out_lavc; - mp_chmap_from_lavc(&in_lavc, in_ch_layout); - mp_chmap_from_lavc(&out_lavc, out_ch_layout); - - if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) { - MP_VERBOSE(af, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc), - mp_chmap_to_str(&out_lavc)); - } - - if (in_lavc.num != map_in.num) { - // For handling NA channels, we would have to add a planarization step. - MP_FATAL(af, "Unsupported channel remapping.\n"); - goto error; - } - - mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc); - transpose_order(s->reorder_in, map_in.num); - - if (mp_chmap_equals(&out_lavc, &map_out)) { - // No intermediate step required - output new format directly. - out_samplefmtp = out_samplefmt; - } else { - // Verify that we really just reorder and/or insert NA channels. - struct mp_chmap withna = out_lavc; - mp_chmap_fill_na(&withna, map_out.num); - if (withna.num != map_out.num) - goto error; - } - mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out); - - s->avrctx_fmt = *out; - mp_audio_set_channels(&s->avrctx_fmt, &out_lavc); - mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp)); - - s->pre_out_fmt = *out; - - // If there are NA channels, the final output will have more channels than - // the avrctx output. Also, avrctx will output planar (out_samplefmtp was - // not overwritten). Allocate the output frame with more channels, so the - // NA channels can be trivially added. - s->pool_fmt = s->avrctx_fmt; - if (map_out.num > out_lavc.num) - mp_audio_set_channels(&s->pool_fmt, &map_out); - - out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout); - - // Real conversion; output is input to avrctx_out. - av_opt_set_int(s->avrctx, "in_channel_layout", in_ch_layout, 0); - av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0); - av_opt_set_int(s->avrctx, "in_sample_rate", s->in_rate, 0); - av_opt_set_int(s->avrctx, "out_sample_rate", s->out_rate, 0); - av_opt_set_int(s->avrctx, "in_sample_fmt", in_samplefmt, 0); - av_opt_set_int(s->avrctx, "out_sample_fmt", out_samplefmtp, 0); - - // Just needs the correct number of channels for deplanarization. - struct mp_chmap fake_chmap; - mp_chmap_set_unknown(&fake_chmap, map_out.num); - uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap); - if (!fake_out_ch_layout) - goto error; - av_opt_set_int(s->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0); - av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0); - - av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmtp, 0); - av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0); - av_opt_set_int(s->avrctx_out, "in_sample_rate", s->out_rate, 0); - av_opt_set_int(s->avrctx_out, "out_sample_rate", s->out_rate, 0); - - // API has weird requirements, quoting avresample.h: - // * This function can only be called when the allocated context is not open. - // * Also, the input channel layout must have already been set. - avresample_set_channel_mapping(s->avrctx, s->reorder_in); - - s->is_resampling = false; - - if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) { - MP_ERR(af, "Cannot open Libavresample Context. \n"); - goto error; - } - return AF_OK; - -error: - close_lavrr(af); - return AF_ERROR; -} - - static int control(struct af_instance *af, int cmd, void *arg) { struct af_resample *s = af->priv; @@ -378,8 +76,12 @@ static int control(struct af_instance *af, int cmd, void *arg) mp_chmap_equals(&in->channels, &orig_in.channels)) ? AF_OK : AF_FALSE; - if (r == AF_OK) - r = configure_lavrr(af, in, out, true); + if (r == AF_OK) { + if (!mp_aconverter_reconfig(s->converter, + in->rate, in->format, in->channels, + out->rate, out->format, out->channels)) + r = AF_ERROR; + } return r; } case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: { @@ -387,17 +89,7 @@ static int control(struct af_instance *af, int cmd, void *arg) return AF_OK; } case AF_CONTROL_RESET: - if (s->avrctx) { -#if HAVE_LIBSWRESAMPLE - swr_close(s->avrctx); - if (swr_init(s->avrctx) < 0) { - close_lavrr(af); - return AF_ERROR; - } -#else - while (avresample_read(s->avrctx, NULL, 1000) > 0) {} -#endif - } + mp_aconverter_flush(s->converter); return AF_OK; } return AF_UNKNOWN; @@ -405,149 +97,40 @@ static int control(struct af_instance *af, int cmd, void *arg) static void uninit(struct af_instance *af) { - close_lavrr(af); -} - -// The LSB is always ignored. -#if BYTE_ORDER == BIG_ENDIAN -#define SHIFT24(x) ((3-(x))*8) -#else -#define SHIFT24(x) (((x)+1)*8) -#endif - -static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa) -{ - for (int p = 0; p < mpa->num_planes; p++) { - void *ptr = mpa->planes[p]; - int total = mpa->samples * mpa->spf; - if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) { - for (int s = 0; s < total; s++) - ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f); - } else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) { - for (int s = 0; s < total; s++) - ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0); - } - } -} - -// This relies on the tricky way mpa was allocated. -static void reorder_planes(struct mp_audio *mpa, int *reorder, - struct mp_chmap *newmap) -{ - struct mp_audio prev = *mpa; - mp_audio_set_channels(mpa, newmap); - - // The trailing planes were never written by avrctx, they're the NA channels. - int next_na = prev.num_planes; + struct af_resample *s = af->priv; - for (int n = 0; n < mpa->num_planes; n++) { - int src = reorder[n]; - assert(src >= -1 && src < prev.num_planes); - if (src >= 0) { - mpa->planes[n] = prev.planes[src]; - } else { - assert(next_na < mpa->num_planes); - mpa->planes[n] = prev.planes[next_na++]; - af_fill_silence(mpa->planes[n], mpa->sstride * mpa->samples, - mpa->format); - } - } + talloc_free(s->converter); } -static int filter_resample(struct af_instance *af, struct mp_audio *in) +static int filter(struct af_instance *af, struct mp_audio *in) { struct af_resample *s = af->priv; - struct mp_audio *out = NULL; - - if (!s->avrctx) - goto error; - int samples = get_out_samples(s, in ? in->samples : 0); - - struct mp_audio out_format = s->pool_fmt; - out = mp_audio_pool_get(af->out_pool, &out_format, samples); - if (!out) - goto error; - if (in) - mp_audio_copy_attributes(out, in); - - if (out->samples) { - out->samples = resample_frame(s->avrctx, out, in); - if (out->samples < 0) - goto error; - } - - struct mp_audio real_out = *out; - mp_audio_copy_config(out, &s->avrctx_fmt); - - if (out->samples && !mp_audio_config_equals(out, &s->pre_out_fmt)) { - assert(af_fmt_is_planar(out->format) && out->format == real_out.format); - reorder_planes(out, s->reorder_out, &s->pool_fmt.channels); - if (!mp_audio_config_equals(out, &s->pre_out_fmt)) { - struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer, - &s->pre_out_fmt, - out->samples); - if (!new) - goto error; - mp_audio_copy_attributes(new, out); - int out_samples = resample_frame(s->avrctx_out, new, out); - talloc_free(out); - out = new; - if (out_samples != new->samples) - goto error; - } - } + mp_aconverter_set_speed(s->converter, s->playback_speed); - extra_output_conversion(af, out); + af->filter_out(af); + struct mp_aframe *aframe = mp_audio_to_aframe(in); + if (!aframe && in) + return -1; talloc_free(in); - if (out->samples) { - af_add_output_frame(af, out); - } else { - talloc_free(out); - } - - af->delay = get_delay(s); + bool ok = mp_aconverter_write_input(s->converter, aframe); + if (!ok) + talloc_free(aframe); - return 0; -error: - talloc_free(in); - talloc_free(out); - return -1; + return ok ? 0 : -1; } -static int filter(struct af_instance *af, struct mp_audio *in) +static int filter_out(struct af_instance *af) { struct af_resample *s = af->priv; - - int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed); - if (s->avrctx && !(!s->is_resampling && new_rate == s->in_rate)) { - AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate, - INT_MAX / 2); - // Essentially, swr/avresample_set_compensation() does 2 things: - // - adjust output sample rate by sample_delta/compensation_distance - // - reset the adjustment after compensation_distance output samples - // Increase the compensation_distance to avoid undesired reset - // semantics - we want to keep the ratio for the whole frame we're - // feeding it, until the next filter() call. - int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1); - r = (AVRational){ r.num * mult, r.den * mult }; - if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) >= 0) { - new_rate = s->in_rate; - s->is_resampling = true; - } - } - - bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01; - if (need_reinit && new_rate != s->in_rate) { - // Before reconfiguring, drain the audio that is still buffered - // in the resampler. - filter_resample(af, NULL); - // Reinitialize resampler. - configure_lavrr(af, &af->fmt_in, &af->fmt_out, false); - } - - return filter_resample(af, in); + bool eof; + struct mp_aframe *out = mp_aconverter_read_output(s->converter, &eof); + if (out) + af_add_output_frame(af, mp_audio_from_aframe(out)); + talloc_free(out); + af->delay = mp_aconverter_get_latency(s->converter); + return 0; } static int af_open(struct af_instance *af) @@ -557,11 +140,9 @@ static int af_open(struct af_instance *af) af->control = control; af->uninit = uninit; af->filter_frame = filter; + af->filter_out = filter_out; - if (s->opts.cutoff <= 0.0) - s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size); - - s->reorder_buffer = mp_audio_pool_create(s); + s->converter = mp_aconverter_create(af->global, af->log, &s->opts); return AF_OK; } @@ -574,12 +155,7 @@ const struct af_info af_info_lavrresample = { .open = af_open, .priv_size = sizeof(struct af_resample), .priv_defaults = &(const struct af_resample) { - .opts = { - .filter_size = 16, - .cutoff = 0.0, - .phase_shift = 10, - .normalize = -1, - }, + .opts = MP_RESAMPLE_OPTS_DEF, .playback_speed = 1.0, .allow_detach = 1, }, @@ -591,7 +167,7 @@ const struct af_info af_info_lavrresample = { OPT_FLAG("detach", allow_detach, 0), OPT_CHOICE("normalize", opts.normalize, 0, ({"no", 0}, {"yes", 1}, {"auto", -1})), - OPT_KEYVALUELIST("o", avopts, 0), + OPT_KEYVALUELIST("o", opts.avopts, 0), {0} }, }; -- cgit v1.2.3