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decoder/wavpack: move 8 and 16 bit conversion to separate functions

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This commit is contained in:
Max Kellermann 2016-12-09 22:49:07 +01:00
parent 61296cedd2
commit ad5abba8c6
1 changed files with 38 additions and 45 deletions
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83
src/decoder/plugins/WavpackDecoderPlugin.cxx
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@ -70,59 +70,45 @@ GetDuration(WavpackContext *wpc)
WavpackGetSampleRate(wpc)); WavpackGetSampleRate(wpc));
} }
/** A pointer type for format converter function. */
typedef void (*format_samples_t)(
int bytes_per_sample,
void *buffer, uint32_t count
);
/* /*
* This function has been borrowed from the tiny player found on * Convert 8 bit.
* wavpack.com. Modifications were required because mpd only handles
* max 24-bit samples.
*/ */
static void static void
format_samples_int(int bytes_per_sample, void *buffer, uint32_t count) format_samples_8(void *buffer, uint32_t count)
{ {
int32_t *src = (int32_t *)buffer; int32_t *src = (int32_t *)buffer;
int8_t *dst = (int8_t *)buffer;
/*
* The asserts like the following one are because we do the
* formatting of samples within a single buffer. The size of
* the output samples never can be greater than the size of
* the input ones. Otherwise we would have an overflow.
*/
static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size");
switch (bytes_per_sample) { /* pass through and align 8-bit samples */
case 1: { std::copy_n(src, count, dst);
int8_t *dst = (int8_t *)buffer; }
/*
* The asserts like the following one are because we do the
* formatting of samples within a single buffer. The size
* of the output samples never can be greater than the size
* of the input ones. Otherwise we would have an overflow.
*/
static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size");
/* pass through and align 8-bit samples */ /*
std::copy_n(src, count, dst); * Convert 16 bit.
break; */
} static void
case 2: { format_samples_16(void *buffer, uint32_t count)
auto *dst = (int16_t *)buffer; {
static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size"); int32_t *src = (int32_t *)buffer;
int16_t *dst = (int16_t *)buffer;
static_assert(sizeof(*dst) <= sizeof(*src), "Wrong size");
/* pass through and align 16-bit samples */ /* pass through and align 16-bit samples */
std::copy_n(src, count, dst); std::copy_n(src, count, dst);
break;
}
case 3:
case 4:
/* do nothing */
break;
}
} }
/* /*
* No conversion necessary. * No conversion necessary.
*/ */
static void static void
format_samples_nop(gcc_unused int bytes_per_sample, gcc_unused void *buffer, format_samples_nop(gcc_unused void *buffer, gcc_unused uint32_t count)
gcc_unused uint32_t count)
{ {
/* do nothing */ /* do nothing */
} }
@ -170,13 +156,21 @@ wavpack_decode(DecoderClient &client, WavpackContext *wpc, bool can_seek)
sample_format, sample_format,
WavpackGetReducedChannels(wpc)); WavpackGetReducedChannels(wpc));
const format_samples_t format_samples = is_float auto *format_samples = format_samples_nop;
? format_samples_nop if (!is_float) {
: format_samples_int; switch (WavpackGetBytesPerSample(wpc)) {
case 1:
format_samples = format_samples_8;
break;
case 2:
format_samples = format_samples_16;
break;
}
}
client.Ready(audio_format, can_seek, GetDuration(wpc)); client.Ready(audio_format, can_seek, GetDuration(wpc));
const int bytes_per_sample = WavpackGetBytesPerSample(wpc);
const int output_sample_size = audio_format.GetFrameSize(); const int output_sample_size = audio_format.GetFrameSize();
/* wavpack gives us all kind of samples in a 32-bit space */ /* wavpack gives us all kind of samples in a 32-bit space */
@ -207,8 +201,7 @@ wavpack_decode(DecoderClient &client, WavpackContext *wpc, bool can_seek)
int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 + int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
0.5); 0.5);
format_samples(bytes_per_sample, chunk, format_samples(chunk, samples_got * audio_format.channels);
samples_got * audio_format.channels);
cmd = client.SubmitData(nullptr, chunk, cmd = client.SubmitData(nullptr, chunk,
samples_got * output_sample_size, samples_got * output_sample_size,