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authorMax Kellermann <max@duempel.org>2013-08-03 21:00:50 +0200
committerMax Kellermann <max@duempel.org>2013-08-03 21:37:56 +0200
commitd1e7b4e38136f9342aad76c685a13adf0e69f869 (patch)
tree49643b937ddfe735511b566a71398da5a945d7aa /src/encoder/FlacEncoderPlugin.cxx
parent67f591a9ce60651da41afc499bd9a22e25314e35 (diff)
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audio_format: convert to C++
Diffstat (limited to '')
-rw-r--r--src/encoder/FlacEncoderPlugin.cxx31
1 files changed, 17 insertions, 14 deletions
diff --git a/src/encoder/FlacEncoderPlugin.cxx b/src/encoder/FlacEncoderPlugin.cxx
index 3aeb96cf7..5e7e7f526 100644
--- a/src/encoder/FlacEncoderPlugin.cxx
+++ b/src/encoder/FlacEncoderPlugin.cxx
@@ -20,7 +20,7 @@
#include "config.h"
#include "FlacEncoderPlugin.hxx"
#include "EncoderAPI.hxx"
-#include "audio_format.h"
+#include "AudioFormat.hxx"
#include "pcm/PcmBuffer.hxx"
#include "util/fifo_buffer.h"
@@ -40,7 +40,7 @@ extern "C" {
struct flac_encoder {
Encoder encoder;
- struct audio_format audio_format;
+ AudioFormat audio_format;
unsigned compression;
FLAC__StreamEncoder *fse;
@@ -160,31 +160,31 @@ flac_encoder_close(Encoder *_encoder)
}
static bool
-flac_encoder_open(Encoder *_encoder, struct audio_format *audio_format,
+flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format,
GError **error)
{
struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
unsigned bits_per_sample;
- encoder->audio_format = *audio_format;
+ encoder->audio_format = audio_format;
/* FIXME: flac should support 32bit as well */
- switch (audio_format->format) {
- case SAMPLE_FORMAT_S8:
+ switch (audio_format.format) {
+ case SampleFormat::S8:
bits_per_sample = 8;
break;
- case SAMPLE_FORMAT_S16:
+ case SampleFormat::S16:
bits_per_sample = 16;
break;
- case SAMPLE_FORMAT_S24_P32:
+ case SampleFormat::S24_P32:
bits_per_sample = 24;
break;
default:
bits_per_sample = 24;
- audio_format->format = SAMPLE_FORMAT_S24_P32;
+ audio_format.format = SampleFormat::S24_P32;
}
/* allocate the encoder */
@@ -263,30 +263,33 @@ flac_encoder_write(Encoder *_encoder,
/* format conversion */
- num_frames = length / audio_format_frame_size(&encoder->audio_format);
+ num_frames = length / encoder->audio_format.GetFrameSize();
num_samples = num_frames * encoder->audio_format.channels;
switch (encoder->audio_format.format) {
- case SAMPLE_FORMAT_S8:
+ case SampleFormat::S8:
exbuffer = encoder->expand_buffer.Get(length * 4);
pcm8_to_flac((int32_t *)exbuffer, (const int8_t *)data,
num_samples);
buffer = exbuffer;
break;
- case SAMPLE_FORMAT_S16:
+ case SampleFormat::S16:
exbuffer = encoder->expand_buffer.Get(length * 2);
pcm16_to_flac((int32_t *)exbuffer, (const int16_t *)data,
num_samples);
buffer = exbuffer;
break;
- case SAMPLE_FORMAT_S24_P32:
- case SAMPLE_FORMAT_S32:
+ case SampleFormat::S24_P32:
+ case SampleFormat::S32:
/* nothing need to be done; format is the same for
both mpd and libFLAC */
buffer = data;
break;
+
+ default:
+ gcc_unreachable();
}
/* feed samples to encoder */