[libav] Things start to settle

src/plugins/score-plugin-gfx/Gfx/Libav/AudioFrameEncoder.hpp

@@ -5,8 +5,6 @@
 extern "C" {
 #include <libavutil/frame.h>
 }
-#define SAMPLE_RATE_TEST 44100
-#define BUFFER_SIZE_TEST 512
 
 #include <ossia/dataflow/sample_to_float.hpp>
 
@@ -15,7 +13,7 @@ namespace Gfx
 
 struct AudioFrameEncoder
 {
-  AudioFrameEncoder(int target_buffer_size)
+  explicit AudioFrameEncoder(int target_buffer_size)
       : target_buffer_size{target_buffer_size}
   {
   }

src/plugins/score-plugin-gfx/Gfx/Libav/LibavEncoder.cpp

@@ -1,12 +1,12 @@
 #include "LibavEncoder.hpp"
 
 #include "LibavOutputStream.hpp"
-
-#include <QDebug>
 extern "C" {
 #include <libswresample/swresample.h>
 }
 
+#include <QDebug>
+
 #if SCORE_HAS_LIBAV
 namespace Gfx
 {
@@ -120,9 +120,10 @@ int LibavEncoder::add_frame(tcb::span<ossia::float_vector> vec)
   auto& stream = streams[audio_stream_index];
   AVFrame* next_frame = stream.get_audio_frame();
 
-  next_frame->sample_rate = SAMPLE_RATE_TEST;
+  next_frame->sample_rate = stream.enc->sample_rate;
   next_frame->format = stream.enc->sample_fmt;
-  next_frame->nb_samples = vec[0].size();
+  next_frame->nb_samples = stream.enc->frame_size;
+  qDebug() << vec[0].size() << next_frame->nb_samples;
   next_frame->ch_layout.nb_channels = channels;
   next_frame->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
 

src/plugins/score-plugin-gfx/Gfx/Libav/LibavOutputStream.hpp

@@ -214,11 +214,13 @@ struct OutputStream
     {
       auto& audio_stgs = score::AppContext().settings<Audio::Settings::Model>();
       nb_samples = audio_stgs.getBufferSize();
-      enc->frame_size = audio_stgs.getBufferSize();
+      enc->frame_size = nb_samples;
+      qDebug() << "Setting frame_size: " << nb_samples;
     }
     else
     {
       nb_samples = enc->frame_size;
+      qDebug() << "Forcing frame_size: " << nb_samples;
     }
     cache_input_frame = alloc_audio_frame(
         enc->sample_fmt, &enc->ch_layout, enc->sample_rate, nb_samples);
@@ -245,7 +247,7 @@ struct OutputStream
       {
         for(int i = 0; i < set.audio_channels; i++)
           this->resamplers.push_back(std::make_unique<r8b::CDSPResampler>(
-              SAMPLE_RATE_TEST, enc->sample_rate, nb_samples, 3.0, 206.91,
+              input_sample_rate, enc->sample_rate, nb_samples * 2, 3.0, 206.91,
               r8b::fprMinPhase));
       }
 
@@ -445,7 +447,7 @@ struct OutputStream
       exit(1);
 
     this->cache_input_frame->pts = this->next_pts;
-    this->next_pts += BUFFER_SIZE_TEST;
+    this->next_pts += this->enc->frame_size;
 
     return this->cache_input_frame;
   }
@@ -499,22 +501,22 @@ struct OutputStream
     return ret == AVERROR_EOF ? 1 : 0;
   }
 
-#define SRC_RATE SAMPLE_RATE_TEST
-#define DST_RATE SAMPLE_RATE_TEST
-  static int64_t conv_audio_pts(SwrContext* ctx, int64_t in, int sample_rate)
-  {
-    //int64_t d = (int64_t) AUDIO_RATE * AUDIO_RATE;
-    int64_t d = (int64_t)sample_rate * sample_rate;
-
-    /* Convert from audio_src_tb to 1/(src_samplerate * dst_samplerate) */
-    in = av_rescale_rnd(in, d, SRC_RATE, AV_ROUND_NEAR_INF);
-
-    /* In units of 1/(src_samplerate * dst_samplerate) */
-    in = swr_next_pts(ctx, in);
-
-    /* Convert from 1/(src_samplerate * dst_samplerate) to audio_dst_tb */
-    return av_rescale_rnd(in, DST_RATE, d, AV_ROUND_NEAR_INF);
-  }
+  // #define SRC_RATE SAMPLE_RATE_TEST
+  // #define DST_RATE SAMPLE_RATE_TEST
+  //   static int64_t conv_audio_pts(SwrContext* ctx, int64_t in, int sample_rate)
+  //   {
+  //     //int64_t d = (int64_t) AUDIO_RATE * AUDIO_RATE;
+  //     int64_t d = (int64_t)sample_rate * sample_rate;
+  //
+  //     /* Convert from audio_src_tb to 1/(src_samplerate * dst_samplerate) */
+  //     in = av_rescale_rnd(in, d, SRC_RATE, AV_ROUND_NEAR_INF);
+  //
+  //     /* In units of 1/(src_samplerate * dst_samplerate) */
+  //     in = swr_next_pts(ctx, in);
+  //
+  //     /* Convert from 1/(src_samplerate * dst_samplerate) to audio_dst_tb */
+  //     return av_rescale_rnd(in, DST_RATE, d, AV_ROUND_NEAR_INF);
+  //   }
 
   int write_audio_frame(AVFormatContext* fmt_ctx, AVFrame* input_frame)
   {