forked from len0rd/rockbox
Streamlined WavPack decoder by utilizing dsp functionality where it was
applicable (like mono conversion and clipping) and eliminating the conversion to 16-bit samples (everything is now returned as 28-bit). This reduced boost ratio (on iRiver) by about 7% on those tracks that require it. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8598 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
parent
f17e35d27b
commit
ff40af42ba
2 changed files with 22 additions and 110 deletions
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@ -675,11 +675,17 @@ static void decorr_mono_pass (struct decorr_pass *dpp, long *buffer, long sample
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// it is clipped and shifted in a single operation. Otherwise, if it's
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// lossless then the last step is to apply the final shift (if any).
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// This function has been modified for RockBox to return all integer samples
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// as 28-bits, and clipping (for lossy mode) has been eliminated because this
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// now happens in the dsp module.
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static void fixup_samples (WavpackStream *wps, long *buffer, ulong sample_count)
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{
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ulong flags = wps->wphdr.flags;
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int shift = (flags & SHIFT_MASK) >> SHIFT_LSB;
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shift += 20 - (flags & BYTES_STORED) * 8; // this provides RockBox with 28-bit data
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if (flags & FLOAT_DATA) {
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float_values (wps, buffer, (flags & MONO_FLAG) ? sample_count : sample_count * 2);
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return;
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@ -689,7 +695,6 @@ static void fixup_samples (WavpackStream *wps, long *buffer, ulong sample_count)
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ulong count = (flags & MONO_FLAG) ? sample_count : sample_count * 2;
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int sent_bits = wps->int32_sent_bits, zeros = wps->int32_zeros;
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int ones = wps->int32_ones, dups = wps->int32_dups;
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// ulong mask = (1 << sent_bits) - 1;
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long *dptr = buffer;
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if (!(flags & HYBRID_FLAG) && !sent_bits && (zeros + ones + dups))
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@ -707,51 +712,22 @@ static void fixup_samples (WavpackStream *wps, long *buffer, ulong sample_count)
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shift += zeros + sent_bits + ones + dups;
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}
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if (flags & HYBRID_FLAG) {
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long min_value, max_value, min_shifted, max_shifted;
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switch (flags & BYTES_STORED) {
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case 0:
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min_shifted = (min_value = -128 >> shift) << shift;
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max_shifted = (max_value = 127 >> shift) << shift;
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break;
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case 1:
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min_shifted = (min_value = -32768 >> shift) << shift;
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max_shifted = (max_value = 32767 >> shift) << shift;
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break;
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case 2:
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min_shifted = (min_value = -8388608 >> shift) << shift;
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max_shifted = (max_value = 8388607 >> shift) << shift;
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break;
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case 3:
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default:
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min_shifted = (min_value = (long) 0x80000000 >> shift) << shift;
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max_shifted = (max_value = (long) 0x7FFFFFFF >> shift) << shift;
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break;
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}
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if (!(flags & MONO_FLAG))
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sample_count *= 2;
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while (sample_count--) {
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if (*buffer < min_value)
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*buffer++ = min_shifted;
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else if (*buffer > max_value)
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*buffer++ = max_shifted;
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else
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*buffer++ <<= shift;
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}
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}
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else if (shift) {
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if (shift > 0) {
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if (!(flags & MONO_FLAG))
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sample_count *= 2;
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while (sample_count--)
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*buffer++ <<= shift;
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}
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else if (shift < 0) {
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shift = -shift;
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if (!(flags & MONO_FLAG))
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sample_count *= 2;
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while (sample_count--)
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*buffer++ >>= shift;
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}
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}
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// This function checks the crc value(s) for an unpacked block, returning the
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@ -24,9 +24,6 @@ CODEC_HEADER
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static struct codec_api *ci;
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#define FORCE_DSP_USE /* fixes some WavPack bugs; adds about 12% to boost ratio
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(when DSP would not have been used) */
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#define BUFFER_SIZE 4096
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static long temp_buffer [BUFFER_SIZE] IBSS_ATTR;
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@ -66,8 +63,7 @@ enum codec_status codec_start(struct codec_api* api)
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(27)); // should be 28...
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next_track:
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@ -79,20 +75,9 @@ enum codec_status codec_start(struct codec_api* api)
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while (!*ci->taginfo_ready && !ci->stop_codec)
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ci->sleep(1);
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#ifdef FORCE_DSP_USE
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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codec_set_replaygain(ci->id3);
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#else
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if (ci->id3->frequency != NATIVE_FREQUENCY ||
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ci->global_settings->replaygain) {
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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codec_set_replaygain(ci->id3);
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}
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else
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ci->configure(CODEC_DSP_ENABLE, (bool *)false);
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#endif
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/* Create a decoder instance */
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wpc = WavpackOpenFileInput (read_callback, error);
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@ -104,6 +89,7 @@ enum codec_status codec_start(struct codec_api* api)
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bps = WavpackGetBytesPerSample (wpc);
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nchans = WavpackGetReducedChannels (wpc);
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ci->configure(DSP_SET_STEREO_MODE, nchans == 2 ? (int *)STEREO_INTERLEAVED : (int *)STEREO_MONO);
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sr_100 = ci->id3->frequency / 100;
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ci->set_elapsed (0);
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@ -141,71 +127,21 @@ enum codec_status codec_start(struct codec_api* api)
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ci->yield ();
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}
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nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / 2);
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nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / nchans);
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if (!nsamples || ci->stop_codec || ci->reload_codec)
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break;
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/* convert mono to stereo here, in place */
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if (nchans == 1) {
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long *dst = temp_buffer + (nsamples * 2);
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long *src = temp_buffer + nsamples;
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long count = nsamples;
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while (count--) {
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*--dst = *--src;
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*--dst = *src;
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if (!(count & 0x7f))
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ci->yield ();
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}
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}
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if (bps == 1) {
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short *dst = (short *) temp_buffer;
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long *src = temp_buffer;
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long count = nsamples;
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while (count--) {
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*dst++ = *src++ << 8;
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*dst++ = *src++ << 8;
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if (!(count & 0x7f))
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ci->yield ();
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}
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}
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else if (bps == 2) {
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short *dst = (short *) temp_buffer;
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long *src = temp_buffer;
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long count = nsamples;
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while (count--) {
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*dst++ = *src++;
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*dst++ = *src++;
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if (!(count & 0x7f))
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ci->yield ();
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}
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}
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else {
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short *dst = (short *) temp_buffer;
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int shift = (bps - 2) * 8;
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long *src = temp_buffer;
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long count = nsamples;
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while (count--) {
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*dst++ = *src++ >> shift;
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*dst++ = *src++ >> shift;
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if (!(count & 0x7f))
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ci->yield ();
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}
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}
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if (ci->stop_codec || ci->reload_codec)
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break;
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while (!ci->pcmbuf_insert ((char *) temp_buffer, nsamples * 4))
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while (!ci->pcmbuf_insert ((char *) temp_buffer, nsamples * nchans * 4))
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ci->sleep (1);
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ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
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ci->yield ();
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}
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if (ci->request_next_track())
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