RepoMirrors
/
mpv
mirror of https://github.com/mpv-player/mpv
1
0
Fork 0
Code Issues Releases Wiki Activity

af_lavrresample: remove unnecessary indirections 

Not sure why struct af_resample_opts even exists. It seems useful to
group the fields set by user options. But storing the current format
conversion parameters doesn't seem very elegant, and having a separate
instance in the "ctx" field isn't helpful either.
This commit is contained in:
wm4 2015-09-08 22:21:19 +02:00
parent 4eae4a5da7
commit af0b903afa
1 changed files with 30 additions and 35 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

65
audio/filter/af_lavrresample.c
View File

@ -66,14 +66,6 @@ struct af_resample_opts {
int linear;
double cutoff;
int normalize;
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;
};
struct af_resample {
@ -85,19 +77,26 @@ struct af_resample {
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 ctx; // opts in the context
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->ctx.in_rate +
avresample_available(s->avrctx) / (double)s->ctx.out_rate;
return avresample_get_delay(s->avrctx) / (double)s->in_rate +
avresample_available(s->avrctx) / (double)s->out_rate;
}
static void drop_all_output(struct af_resample *s)
{
@ -110,7 +109,7 @@ static int get_out_samples(struct af_resample *s, int in_samples)
#else
static double get_delay(struct af_resample *s)
{
int64_t base = s->ctx.in_rate * (int64_t)s->ctx.out_rate;
int64_t base = s->in_rate * (int64_t)s->out_rate;
return swr_get_delay(s->avrctx, base) / (double)base;
}
static void drop_all_output(struct af_resample *s)
@ -122,8 +121,8 @@ static int get_out_samples(struct af_resample *s, int in_samples)
#if LIBSWRESAMPLE_VERSION_MAJOR > 1 || LIBSWRESAMPLE_VERSION_MINOR >= 2
return swr_get_out_samples(s->avrctx, in_samples);
#else
return av_rescale_rnd(in_samples, s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP)
+ swr_get_delay(s->avrctx, s->ctx.out_rate);
return av_rescale_rnd(in_samples, s->out_rate, s->in_rate, AV_ROUND_UP)
+ swr_get_delay(s->avrctx, s->out_rate);
#endif
}
#endif
@ -216,23 +215,19 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
out_samplefmtp == AV_SAMPLE_FMT_NONE)
goto error;
s->ctx.out_rate = out->rate;
s->ctx.in_rate_af = in->rate;
s->ctx.in_rate = rate_from_speed(in->rate, s->playback_speed);
s->ctx.out_format = out->format;
s->ctx.in_format = in->format;
s->ctx.out_channels= out->channels;
s->ctx.in_channels = in->channels;
s->ctx.filter_size = s->opts.filter_size;
s->ctx.phase_shift = s->opts.phase_shift;
s->ctx.linear = s->opts.linear;
s->ctx.cutoff = s->opts.cutoff;
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->ctx.filter_size, 0);
av_opt_set_int(s->avrctx, "phase_shift", s->ctx.phase_shift, 0);
av_opt_set_int(s->avrctx, "linear_interp", s->ctx.linear, 0);
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->ctx.cutoff, 0);
av_opt_set_double(s->avrctx, "cutoff", s->opts.cutoff, 0);
#if HAVE_LIBSWRESAMPLE
av_opt_set_double(s->avrctx, "rematrix_maxval", s->opts.normalize ? 1 : 1000, 0);
@ -304,8 +299,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
// 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->ctx.in_rate, 0);
av_opt_set_int(s->avrctx, "out_sample_rate", s->ctx.out_rate, 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);
@ -319,8 +314,8 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
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->ctx.out_rate, 0);
av_opt_set_int(s->avrctx_out, "out_sample_rate", s->ctx.out_rate, 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.
@ -391,8 +386,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
return AF_OK;
case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
s->playback_speed = *(double *)arg;
int new_rate = rate_from_speed(s->ctx.in_rate_af, s->playback_speed);
if (new_rate != s->ctx.in_rate && s->avrctx && af->fmt_out.format) {
int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
if (new_rate != s->in_rate && s->avrctx && af->fmt_out.format) {
// Before reconfiguring, drain the audio that is still buffered
// in the resampler.
af->filter_frame(af, NULL);