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mp3: send 24 bit PCM data

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libmad produces samples of more than 24 bit.  Rounding that down to 16
bits using dithering makes those people lose quality who have a 24 bit
capable sound device.  Send 24 bit PCM data, and let the receiver
decide whether to apply 16 bit dithering.
This commit is contained in:
Max Kellermann 2008-10-23 16:58:38 +02:00
parent bf5774edbd
commit 2cc2420f8c
1 changed files with 24 additions and 63 deletions
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87
src/inputPlugins/mp3_plugin.c
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@ -51,77 +51,39 @@ enum muteframe {
static int gaplessPlaybackEnabled;
/* this is stolen from mpg321! */
struct audio_dither {
mad_fixed_t error[3];
mad_fixed_t random;
};
static unsigned long prng(unsigned long state)
static inline int32_t
mad_fixed_to_24_sample(mad_fixed_t sample)
{
return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}
static int16_t audio_linear_dither(mad_fixed_t sample,
struct audio_dither *dither)
{
mad_fixed_t output, mask, rnd;
enum {
bits = 16,
scalebits = MAD_F_FRACBITS + 1 - bits,
bits = 24,
MIN = -MAD_F_ONE,
MAX = MAD_F_ONE - 1
};
sample += dither->error[0] - dither->error[1] + dither->error[2];
/* round */
sample = sample + (1L << (MAD_F_FRACBITS - bits));
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0] / 2;
/* clip */
if (sample > MAX)
sample = MAX;
else if (sample < MIN)
sample = MIN;
output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
mask = (1L << scalebits) - 1;
rnd = prng(dither->random);
output += (rnd & mask) - (dither->random & mask);
dither->random = rnd;
if (output > MAX) {
output = MAX;
if (sample > MAX)
sample = MAX;
} else if (output < MIN) {
output = MIN;
if (sample < MIN)
sample = MIN;
}
output &= ~mask;
dither->error[0] = sample - output;
return (int16_t)(output >> scalebits);
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - bits);
}
static unsigned dither_buffer(int16_t *dest0, const struct mad_synth *synth,
struct audio_dither *dither,
unsigned int start, unsigned int end,
unsigned int num_channels)
static void
mad_fixed_to_24_buffer(int32_t *dest, const struct mad_synth *synth,
unsigned int start, unsigned int end,
unsigned int num_channels)
{
int16_t *dest = dest0;
unsigned int i, c;
for (i = start; i < end; ++i) {
for (c = 0; c < num_channels; ++c)
*dest++ = audio_linear_dither(synth->pcm.samples[c][i],
dither);
*dest++ = mad_fixed_to_24_sample(synth->pcm.samples[c][i]);
}
return dest - dest0;
}
/* end of stolen stuff from mpg321 */
@ -145,7 +107,7 @@ typedef struct _mp3DecodeData {
struct mad_synth synth;
mad_timer_t timer;
unsigned char readBuffer[READ_BUFFER_SIZE];
int16_t outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
int32_t outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
float totalTime;
float elapsedTime;
enum muteframe muteFrame;
@ -164,7 +126,6 @@ typedef struct _mp3DecodeData {
unsigned long bitRate;
struct decoder *decoder;
InputStream *inStream;
struct audio_dither dither;
enum mad_layer layer;
} mp3DecodeData;
@ -187,7 +148,6 @@ static void initMp3DecodeData(mp3DecodeData * data, struct decoder *decoder,
data->decoder = decoder;
data->inStream = inStream;
data->layer = 0;
memset(&(data->dither), 0, sizeof(struct audio_dither));
mad_stream_init(&data->stream);
mad_stream_options(&data->stream, MAD_OPTION_IGNORECRC);
@ -933,10 +893,11 @@ mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
i += num_samples;
num_samples = dither_buffer(data->outputBuffer,
&data->synth, &data->dither,
i - num_samples, i,
MAD_NCHANNELS(&(data->frame).header));
mad_fixed_to_24_buffer(data->outputBuffer,
&data->synth,
i - num_samples, i,
MAD_NCHANNELS(&(data->frame).header));
num_samples *= MAD_NCHANNELS(&(data->frame).header);
cmd = decoder_data(decoder, data->inStream,
data->inStream->seekable,
@ -1022,7 +983,7 @@ mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
struct audio_format * af)
{
af->bits = 16;
af->bits = 24;
af->sample_rate = (data->frame).header.samplerate;
af->channels = MAD_NCHANNELS(&(data->frame).header);
}