subject:"\[FFmpeg\-devel\] \[PATCH\] avfilter\: add square audio source filter"

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

2019-07-27 Thread Nicolas George

Paul B Mahol (12019-07-14):
> Signed-off-by: Paul B Mahol 
> ---
>  doc/filters.texi | 44 
>  libavfilter/Makefile |  1 +
>  libavfilter/allfilters.c |  1 +
>  libavfilter/asrc_sine.c  | 86 ++--
>  4 files changed, 119 insertions(+), 13 deletions(-)
> 
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 3108ad349e..b94eddefe5 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -5895,6 +5895,50 @@ sine=1000:samples_per_frame='st(0,mod(n,5)); 
> 1602-not(not(eq(ld(0),1)+eq(ld(0),3
>  @end example
>  @end itemize
>  
> +@section square
> +
> +Generate an audio signal made of a square wave with custom amplitude.
> +
> +The audio signal is bit-exact.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item amplitude, a
> +Set the carrier amplitude, Default is 0.2.
> +
> +@item frequency, f
> +Set the carrier frequency. Default is 440 Hz.
> +
> +@item sample_rate, r
> +Specify the sample rate, default is 44100.
> +
> +@item duration, d
> +Specify the duration of the generated audio stream.
> +
> +@item samples_per_frame
> +Set the number of samples per output frame.
> +
> +The expression can contain the following constants:
> +
> +@table @option
> +@item n
> +The (sequential) number of the output audio frame, starting from 0.
> +
> +@item pts
> +The PTS (Presentation TimeStamp) of the output audio frame,
> +expressed in @var{TB} units.
> +
> +@item t
> +The PTS of the output audio frame, expressed in seconds.
> +
> +@item TB
> +The timebase of the output audio frames.
> +@end table
> +
> +Default is @code{1024}.
> +@end table
> +
>  @c man end AUDIO SOURCES
>  
>  @chapter Audio Sinks
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 455c809b15..b958450a80 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -148,6 +148,7 @@ OBJS-$(CONFIG_FLITE_FILTER)  += 
> asrc_flite.o
>  OBJS-$(CONFIG_HILBERT_FILTER)+= asrc_hilbert.o
>  OBJS-$(CONFIG_SINC_FILTER)   += asrc_sinc.o
>  OBJS-$(CONFIG_SINE_FILTER)   += asrc_sine.o
> +OBJS-$(CONFIG_SQUARE_FILTER) += asrc_sine.o
>  
>  OBJS-$(CONFIG_ANULLSINK_FILTER)  += asink_anullsink.o
>  
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 04a3df7d56..510e0d65dd 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -140,6 +140,7 @@ extern AVFilter ff_asrc_flite;
>  extern AVFilter ff_asrc_hilbert;
>  extern AVFilter ff_asrc_sinc;
>  extern AVFilter ff_asrc_sine;
> +extern AVFilter ff_asrc_square;
>  
>  extern AVFilter ff_asink_anullsink;
>  
> diff --git a/libavfilter/asrc_sine.c b/libavfilter/asrc_sine.c
> index 3a87210b4b..e3153f03e4 100644
> --- a/libavfilter/asrc_sine.c
> +++ b/libavfilter/asrc_sine.c
> @@ -30,6 +30,7 @@
>  
>  typedef struct SineContext {
>  const AVClass *class;
> +double amplitude;
>  double frequency;
>  double beep_factor;
>  char *samples_per_frame;
> @@ -38,6 +39,7 @@ typedef struct SineContext {
>  int64_t duration;
>  int16_t *sin;
>  int64_t pts;

> +int square_amplitude;

Only used locally, do not put it in the context.

>  uint32_t phi;  ///< current phase of the sine (2pi = 1<<32)
>  uint32_t dphi; ///< phase increment between two samples
>  unsigned beep_period;
> @@ -45,6 +47,8 @@ typedef struct SineContext {
>  unsigned beep_length;
>  uint32_t phi_beep;  ///< current phase of the beep
>  uint32_t dphi_beep; ///< phase increment of the beep
> +

> +void (*generate)(struct SineContext *sine, AVFrame *frame);

Ditto. There is really no need for a function pointer at all.

>  } SineContext;
>  
>  #define CONTEXT SineContext
> @@ -141,11 +145,51 @@ enum {
>  VAR_VARS_NB
>  };
>  
> +static void generate_sine(SineContext *sine, AVFrame *frame)
> +{
> +int16_t *samples = (int16_t *)frame->data[0];
> +
> +for (int i = 0; i < frame->nb_samples; i++) {
> +samples[i] = sine->sin[sine->phi >> (32 - LOG_PERIOD)];
> +sine->phi += sine->dphi;
> +if (sine->beep_index < sine->beep_length) {
> +samples[i] += sine->sin[sine->phi_beep >> (32 - LOG_PERIOD)] << 
> 1;
> +sine->phi_beep += sine->dphi_beep;
> +}
> +if (++sine->beep_index == sine->beep_period)
> +sine->beep_index = 0;
> +}
> +}
> +

> +static void generate_square(SineContext *sine, AVFrame *frame)
> +{
> +int16_t *samples = (int16_t *)frame->data[0];
> +
> +for (int i = 0; i < frame->nb_samples; i++) {
> +int16_t sample = sine->sin[sine->phi >> (32 - LOG_PERIOD)];
> +
> +if (sample >= 0)
> +sample = sine->square_amplitude;
> +else
> +sample = -sine->square_amplitude;
> +
> +samples[i] = sample;
> +sine->phi += sine->dphi;
> +}
> +}

This looks like a needlessly complicated way of generating a

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

2019-07-27 Thread Nicolas George

Paul B Mahol (12019-07-26):
> Nicolas, are you fine with this changes?

Sorry, my mail server had trouble recently. On it.

Regards,

-- 
  Nicolas George


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Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

2019-07-26 Thread Paul B Mahol

Nicolas, are you fine with this changes?

On Wed, Jul 24, 2019 at 1:29 PM Paul B Mahol  wrote:

> On 7/14/19, Paul B Mahol  wrote:
> > Signed-off-by: Paul B Mahol 
> > ---
> >  doc/filters.texi | 44 
> >  libavfilter/Makefile |  1 +
> >  libavfilter/allfilters.c |  1 +
> >  libavfilter/asrc_sine.c  | 86 ++--
> >  4 files changed, 119 insertions(+), 13 deletions(-)
> >
>
> Will apply ASAP!
>
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Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

2019-07-24 Thread Paul B Mahol

On 7/14/19, Paul B Mahol  wrote:
> Signed-off-by: Paul B Mahol 
> ---
>  doc/filters.texi | 44 
>  libavfilter/Makefile |  1 +
>  libavfilter/allfilters.c |  1 +
>  libavfilter/asrc_sine.c  | 86 ++--
>  4 files changed, 119 insertions(+), 13 deletions(-)
>

Will apply ASAP!
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[FFmpeg-devel] [PATCH] avfilter: add square audio source filter

2019-07-14 Thread Paul B Mahol

Signed-off-by: Paul B Mahol 
---
 doc/filters.texi | 44 
 libavfilter/Makefile |  1 +
 libavfilter/allfilters.c |  1 +
 libavfilter/asrc_sine.c  | 86 ++--
 4 files changed, 119 insertions(+), 13 deletions(-)

diff --git a/doc/filters.texi b/doc/filters.texi
index 3108ad349e..b94eddefe5 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -5895,6 +5895,50 @@ sine=1000:samples_per_frame='st(0,mod(n,5)); 
1602-not(not(eq(ld(0),1)+eq(ld(0),3
 @end example
 @end itemize
 
+@section square
+
+Generate an audio signal made of a square wave with custom amplitude.
+
+The audio signal is bit-exact.
+
+The filter accepts the following options:
+
+@table @option
+@item amplitude, a
+Set the carrier amplitude, Default is 0.2.
+
+@item frequency, f
+Set the carrier frequency. Default is 440 Hz.
+
+@item sample_rate, r
+Specify the sample rate, default is 44100.
+
+@item duration, d
+Specify the duration of the generated audio stream.
+
+@item samples_per_frame
+Set the number of samples per output frame.
+
+The expression can contain the following constants:
+
+@table @option
+@item n
+The (sequential) number of the output audio frame, starting from 0.
+
+@item pts
+The PTS (Presentation TimeStamp) of the output audio frame,
+expressed in @var{TB} units.
+
+@item t
+The PTS of the output audio frame, expressed in seconds.
+
+@item TB
+The timebase of the output audio frames.
+@end table
+
+Default is @code{1024}.
+@end table
+
 @c man end AUDIO SOURCES
 
 @chapter Audio Sinks
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 455c809b15..b958450a80 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -148,6 +148,7 @@ OBJS-$(CONFIG_FLITE_FILTER)  += asrc_flite.o
 OBJS-$(CONFIG_HILBERT_FILTER)+= asrc_hilbert.o
 OBJS-$(CONFIG_SINC_FILTER)   += asrc_sinc.o
 OBJS-$(CONFIG_SINE_FILTER)   += asrc_sine.o
+OBJS-$(CONFIG_SQUARE_FILTER) += asrc_sine.o
 
 OBJS-$(CONFIG_ANULLSINK_FILTER)  += asink_anullsink.o
 
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 04a3df7d56..510e0d65dd 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -140,6 +140,7 @@ extern AVFilter ff_asrc_flite;
 extern AVFilter ff_asrc_hilbert;
 extern AVFilter ff_asrc_sinc;
 extern AVFilter ff_asrc_sine;
+extern AVFilter ff_asrc_square;
 
 extern AVFilter ff_asink_anullsink;
 
diff --git a/libavfilter/asrc_sine.c b/libavfilter/asrc_sine.c
index 3a87210b4b..e3153f03e4 100644
--- a/libavfilter/asrc_sine.c
+++ b/libavfilter/asrc_sine.c
@@ -30,6 +30,7 @@
 
 typedef struct SineContext {
 const AVClass *class;
+double amplitude;
 double frequency;
 double beep_factor;
 char *samples_per_frame;
@@ -38,6 +39,7 @@ typedef struct SineContext {
 int64_t duration;
 int16_t *sin;
 int64_t pts;
+int square_amplitude;
 uint32_t phi;  ///< current phase of the sine (2pi = 1<<32)
 uint32_t dphi; ///< phase increment between two samples
 unsigned beep_period;
@@ -45,6 +47,8 @@ typedef struct SineContext {
 unsigned beep_length;
 uint32_t phi_beep;  ///< current phase of the beep
 uint32_t dphi_beep; ///< phase increment of the beep
+
+void (*generate)(struct SineContext *sine, AVFrame *frame);
 } SineContext;
 
 #define CONTEXT SineContext
@@ -141,11 +145,51 @@ enum {
 VAR_VARS_NB
 };
 
+static void generate_sine(SineContext *sine, AVFrame *frame)
+{
+int16_t *samples = (int16_t *)frame->data[0];
+
+for (int i = 0; i < frame->nb_samples; i++) {
+samples[i] = sine->sin[sine->phi >> (32 - LOG_PERIOD)];
+sine->phi += sine->dphi;
+if (sine->beep_index < sine->beep_length) {
+samples[i] += sine->sin[sine->phi_beep >> (32 - LOG_PERIOD)] << 1;
+sine->phi_beep += sine->dphi_beep;
+}
+if (++sine->beep_index == sine->beep_period)
+sine->beep_index = 0;
+}
+}
+
+static void generate_square(SineContext *sine, AVFrame *frame)
+{
+int16_t *samples = (int16_t *)frame->data[0];
+
+for (int i = 0; i < frame->nb_samples; i++) {
+int16_t sample = sine->sin[sine->phi >> (32 - LOG_PERIOD)];
+
+if (sample >= 0)
+sample = sine->square_amplitude;
+else
+sample = -sine->square_amplitude;
+
+samples[i] = sample;
+sine->phi += sine->dphi;
+}
+}
+
 static av_cold int init(AVFilterContext *ctx)
 {
 int ret;
 SineContext *sine = ctx->priv;
 
+if (!strcmp(ctx->filter->name, "square")) {
+sine->square_amplitude = -sine->amplitude * INT16_MIN;
+sine->generate = generate_square;
+} else {
+sine->generate = generate_sine;
+}
+
 if (!(sine->sin = av_malloc(sizeof(*sine->sin) << LOG_PERIOD)))
 return AVERROR(ENOMEM);
 sine->dphi = ldexp(sine->frequency, 32) / sine->sample_rate + 0.5;
@@ -224,8 +268,7 @@

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

Re: [FFmpeg-devel] [PATCH] avfilter: add square audio source filter

[FFmpeg-devel] [PATCH] avfilter: add square audio source filter

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