Fix a memory leak in audio encoding.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index bc349bef275304a57533281667cc5a892b80feee..f044ecd2f743c24855eefb474cfb2a158341fb01 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -66,7 +66,7 @@ void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src
 }  // namespace
 
 Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
-       : httpd("test.ts", WIDTH, HEIGHT),
+       : httpd(LOCAL_DUMP_FILE_NAME, WIDTH, HEIGHT),
          num_cards(num_cards),
          mixer_surface(create_surface(format)),
          h264_encoder_surface(create_surface(format)),
@@ -157,6 +157,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
        // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
        // and there's a limit to how important the peak meter is.
        peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16);
+
+       alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
 }
 
 Mixer::~Mixer()
@@ -172,6 +174,8 @@ Mixer::~Mixer()
                }
                cards[card_index].usb->stop_dequeue_thread();
        }
+
+       h264_encoder.reset(nullptr);
 }
 
 namespace {
@@ -578,22 +582,9 @@ void Mixer::process_audio_one_frame()
 
 //     float limiter_att, compressor_att;
 
-       // Then a limiter at +0 dB (so, -14 dBFS) to take out the worst peaks only.
-       // Note that since ratio is not infinite, we could go slightly higher than this.
-       // Probably more tuning is warranted here.
-       {
-               float threshold = pow(10.0f, (ref_level_dbfs + 0.0f) / 20.0f);  // +0 dB.
-               float ratio = 30.0f;
-               float attack_time = 0.0f;  // Instant.
-               float release_time = 0.005f;
-               float makeup_gain = 1.0f;  // 0 dB.
-               limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-//             limiter_att = limiter.get_attenuation();
-       }
-
-       // Finally, the real compressor.
-       {
-               float threshold = pow(10.0f, (ref_level_dbfs - 12.0f) / 20.0f);  // -12 dB.
+       // The real compressor.
+       if (compressor_enabled) {
+               float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
                float ratio = 20.0f;
                float attack_time = 0.005f;
                float release_time = 0.040f;
@@ -602,6 +593,18 @@ void Mixer::process_audio_one_frame()
 //             compressor_att = compressor.get_attenuation();
        }
 
+       // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
+       // Note that since ratio is not infinite, we could go slightly higher than this.
+       if (limiter_enabled) {
+               float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
+               float ratio = 30.0f;
+               float attack_time = 0.0f;  // Instant.
+               float release_time = 0.020f;
+               float makeup_gain = 1.0f;  // 0 dB.
+               limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+//             limiter_att = limiter.get_attenuation();
+       }
+
 //     printf("limiter=%+5.1f  compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
 
        // Upsample 4x to find interpolated peak.
@@ -624,7 +627,12 @@ void Mixer::process_audio_one_frame()
        float *ptrs[] = { left.data(), right.data() };
        r128.process(left.size(), ptrs);
 
-       // Actually add the samples to the output.
+       // Send the samples to the sound card.
+       if (alsa) {
+               alsa->write(samples_out);
+       }
+
+       // And finally add them to the output.
        h264_encoder->add_audio(pts_int, move(samples_out));
 }