Fix a small theoretical inefficiency; does not actually matter in practice, just...

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index 59b5a70322e5cd249e0503ec7b11c31b0934868b..13eed33048fe5fcf659bf1ed762bb522ef4adfcc 100644 (file)
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -264,8 +264,7 @@ bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
        assert(num_channels > 0);
 
        // Convert the audio to fp32.
-       vector<float> audio;
-       audio.resize(num_samples * num_channels);
+       unique_ptr<float[]> audio(new float[num_samples * num_channels]);
        unsigned channel_index = 0;
        for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
                switch (audio_format.bits_per_sample) {
@@ -273,13 +272,13 @@ bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
                        assert(num_samples == 0);
                        break;
                case 16:
-                       convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+                       convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
                        break;
                case 24:
-                       convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+                       convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
                        break;
                case 32:
-                       convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+                       convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
                        break;
                default:
                        fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
@@ -289,7 +288,7 @@ bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
 
        // Now add it.
        int64_t local_pts = device->next_local_pts;
-       device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+       device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
        device->next_local_pts = local_pts + frame_length;
        return true;
 }