Add support for 16-bit audio.

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index a9b2d4c0f87a4b4ced15c1bc58d4387e8136893e..a534aace6b656196df8343062e203ae00435f339 100644 (file)
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -19,6 +19,24 @@ namespace {
 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
 // (usually including multiple channels at a time).
 
+void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
+                             const uint8_t *src, size_t in_channel, size_t in_num_channels,
+                             size_t num_samples)
+{
+       assert(in_channel < in_num_channels);
+       assert(out_channel < out_num_channels);
+       src += in_channel * 2;
+       dst += out_channel;
+
+       for (size_t i = 0; i < num_samples; ++i) {
+               int16_t s = le16toh(*(int16_t *)src);
+               *dst = s * (1.0f / 32768.0f);
+
+               src += 2 * in_num_channels;
+               dst += out_num_channels;
+       }
+}
+
 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
                              size_t num_samples)
@@ -120,7 +138,7 @@ void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
        unsigned num_channels = device->interesting_channels.size();
        assert(num_channels > 0);
 
-       // Convert the audio to stereo fp32.
+       // Convert the audio to fp32.
        vector<float> audio;
        audio.resize(num_samples * num_channels);
        unsigned channel_index = 0;
@@ -129,6 +147,9 @@ void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned
                case 0:
                        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);
+                       break;
                case 24:
                        convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
                        break;