diff mbox series

[v2,02/17] audio: change type and name of the resample buffer

Message ID 20230206185237.8358-2-vr_qemu@t-online.de (mailing list archive)
State New, archived
Headers show
Series audio: improve callback interface for audio frontends | expand

Commit Message

Volker Rümelin Feb. 6, 2023, 6:52 p.m. UTC 
Change the type of the resample buffer from struct st_sample *
to STSampleBuffer. Also change the name from buf to resample_buf
for better readability.

The new variables resample_buf.size and resample_buf.pos will be
used after the next patches. There is no functional change.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c          | 15 ++++++++-------
 audio/audio_int.h      |  4 ++--
 audio/audio_template.h | 10 ++++++----
 3 files changed, 16 insertions(+), 13 deletions(-)

Comments

Marc-André Lureau Feb. 22, 2023, 10:49 a.m. UTC | #1 
On Mon, Feb 6, 2023 at 10:52 PM Volker Rümelin <vr_qemu@t-online.de> wrote:
>
> Change the type of the resample buffer from struct st_sample *
> to STSampleBuffer. Also change the name from buf to resample_buf
> for better readability.
>
> The new variables resample_buf.size and resample_buf.pos will be
> used after the next patches. There is no functional change.
>
> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>

Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>


> ---
>  audio/audio.c          | 15 ++++++++-------
>  audio/audio_int.h      |  4 ++--
>  audio/audio_template.h | 10 ++++++----
>  3 files changed, 16 insertions(+), 13 deletions(-)
>
> diff --git a/audio/audio.c b/audio/audio.c
> index a0b54e4a2e..a399147486 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -555,7 +555,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
>  {
>      HWVoiceIn *hw = sw->hw;
>      size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
> -    struct st_sample *src, *dst = sw->buf;
> +    struct st_sample *src, *dst = sw->resample_buf.buffer;
>
>      live = hw->total_samples_captured - sw->total_hw_samples_acquired;
>      if (!live) {
> @@ -595,10 +595,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
>      }
>
>      if (!hw->pcm_ops->volume_in) {
> -        mixeng_volume (sw->buf, ret, &sw->vol);
> +        mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
>      }
>
> -    sw->clip (buf, sw->buf, ret);
> +    sw->clip(buf, sw->resample_buf.buffer, ret);
>      sw->total_hw_samples_acquired += total;
>      return ret * sw->info.bytes_per_frame;
>  }
> @@ -706,10 +706,10 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
>      samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
>      samples = MIN(samples, size / sw->info.bytes_per_frame);
>      if (samples) {
> -        sw->conv(sw->buf, buf, samples);
> +        sw->conv(sw->resample_buf.buffer, buf, samples);
>
>          if (!sw->hw->pcm_ops->volume_out) {
> -            mixeng_volume(sw->buf, samples, &sw->vol);
> +            mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol);
>          }
>      }
>
> @@ -724,7 +724,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
>          osamp = blck;
>          st_rate_flow_mix (
>              sw->rate,
> -            sw->buf + pos,
> +            sw->resample_buf.buffer + pos,
>              sw->hw->mix_buf.buffer + wpos,
>              &isamp,
>              &osamp
> @@ -1061,7 +1061,8 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
>                  size_t bytes = to_write * hw->info.bytes_per_frame;
>                  size_t written;
>
> -                sw->buf = hw->mix_buf.buffer + rpos2;
> +                sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
> +                sw->resample_buf.size = to_write;
>                  written = audio_pcm_sw_write (sw, NULL, bytes);
>                  if (written - bytes) {
>                      dolog("Could not mix %zu bytes into a capture "
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index 061845dcc2..8b163e1759 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -109,7 +109,7 @@ struct SWVoiceOut {
>      struct audio_pcm_info info;
>      t_sample *conv;
>      int64_t ratio;
> -    struct st_sample *buf;
> +    STSampleBuffer resample_buf;
>      void *rate;
>      size_t total_hw_samples_mixed;
>      int active;
> @@ -129,7 +129,7 @@ struct SWVoiceIn {
>      int64_t ratio;
>      void *rate;
>      size_t total_hw_samples_acquired;
> -    struct st_sample *buf;
> +    STSampleBuffer resample_buf;
>      f_sample *clip;
>      HWVoiceIn *hw;
>      char *name;
> diff --git a/audio/audio_template.h b/audio/audio_template.h
> index dd87170cbd..a0b653f52c 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -95,13 +95,13 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
>
>  static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
>  {
> -    g_free (sw->buf);
> +    g_free(sw->resample_buf.buffer);
> +    sw->resample_buf.buffer = NULL;
> +    sw->resample_buf.size = 0;
>
>      if (sw->rate) {
>          st_rate_stop (sw->rate);
>      }
> -
> -    sw->buf = NULL;
>      sw->rate = NULL;
>  }
>
> @@ -138,7 +138,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
>          return -1;
>      }
>
> -    sw->buf = g_new0(st_sample, samples);
> +    sw->resample_buf.buffer = g_new0(st_sample, samples);
> +    sw->resample_buf.size = samples;
> +    sw->resample_buf.pos = 0;
>
>  #ifdef DAC
>      sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
> --
> 2.35.3
>


--
Marc-André Lureau
diff mbox series

Patch

diff --git a/audio/audio.c b/audio/audio.c
index a0b54e4a2e..a399147486 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -555,7 +555,7 @@  static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 {
     HWVoiceIn *hw = sw->hw;
     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
-    struct st_sample *src, *dst = sw->buf;
+    struct st_sample *src, *dst = sw->resample_buf.buffer;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
@@ -595,10 +595,10 @@  static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
     }
 
     if (!hw->pcm_ops->volume_in) {
-        mixeng_volume (sw->buf, ret, &sw->vol);
+        mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
     }
 
-    sw->clip (buf, sw->buf, ret);
+    sw->clip(buf, sw->resample_buf.buffer, ret);
     sw->total_hw_samples_acquired += total;
     return ret * sw->info.bytes_per_frame;
 }
@@ -706,10 +706,10 @@  static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
     samples = MIN(samples, size / sw->info.bytes_per_frame);
     if (samples) {
-        sw->conv(sw->buf, buf, samples);
+        sw->conv(sw->resample_buf.buffer, buf, samples);
 
         if (!sw->hw->pcm_ops->volume_out) {
-            mixeng_volume(sw->buf, samples, &sw->vol);
+            mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol);
         }
     }
 
@@ -724,7 +724,7 @@  static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         osamp = blck;
         st_rate_flow_mix (
             sw->rate,
-            sw->buf + pos,
+            sw->resample_buf.buffer + pos,
             sw->hw->mix_buf.buffer + wpos,
             &isamp,
             &osamp
@@ -1061,7 +1061,8 @@  static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
                 size_t bytes = to_write * hw->info.bytes_per_frame;
                 size_t written;
 
-                sw->buf = hw->mix_buf.buffer + rpos2;
+                sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
+                sw->resample_buf.size = to_write;
                 written = audio_pcm_sw_write (sw, NULL, bytes);
                 if (written - bytes) {
                     dolog("Could not mix %zu bytes into a capture "
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 061845dcc2..8b163e1759 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -109,7 +109,7 @@  struct SWVoiceOut {
     struct audio_pcm_info info;
     t_sample *conv;
     int64_t ratio;
-    struct st_sample *buf;
+    STSampleBuffer resample_buf;
     void *rate;
     size_t total_hw_samples_mixed;
     int active;
@@ -129,7 +129,7 @@  struct SWVoiceIn {
     int64_t ratio;
     void *rate;
     size_t total_hw_samples_acquired;
-    struct st_sample *buf;
+    STSampleBuffer resample_buf;
     f_sample *clip;
     HWVoiceIn *hw;
     char *name;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index dd87170cbd..a0b653f52c 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -95,13 +95,13 @@  static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
 
 static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
 {
-    g_free (sw->buf);
+    g_free(sw->resample_buf.buffer);
+    sw->resample_buf.buffer = NULL;
+    sw->resample_buf.size = 0;
 
     if (sw->rate) {
         st_rate_stop (sw->rate);
     }
-
-    sw->buf = NULL;
     sw->rate = NULL;
 }
 
@@ -138,7 +138,9 @@  static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
         return -1;
     }
 
-    sw->buf = g_new0(st_sample, samples);
+    sw->resample_buf.buffer = g_new0(st_sample, samples);
+    sw->resample_buf.size = samples;
+    sw->resample_buf.pos = 0;
 
 #ifdef DAC
     sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);