From 02cc77cd821a04718a9dfb84134c9520859c3305 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Fri, 22 Aug 2014 06:56:58 +0200 Subject: decoder/dsf: fix multi-channel files The plugin was horribly bugged for files that were not stereo. --- NEWS | 1 + src/decoder/plugins/DsfDecoderPlugin.cxx | 56 ++++++++++++++++++++++++++------ 2 files changed, 47 insertions(+), 10 deletions(-) diff --git a/NEWS b/NEWS index 9c9e02eae..ca9653f55 100644 --- a/NEWS +++ b/NEWS @@ -43,6 +43,7 @@ ver 0.19 (not yet released) - audiofile: log libaudiofile errors - dsdiff, dsf: report bit rate - dsf: support DSD512 + - dsf: support multi-channel files - dsf: fix noise at end of malformed file - sndfile: support scanning remote files - sndfile: support tags "comment", "album", "track", "genre" diff --git a/src/decoder/plugins/DsfDecoderPlugin.cxx b/src/decoder/plugins/DsfDecoderPlugin.cxx index 8c6c3a03d..953096a45 100644 --- a/src/decoder/plugins/DsfDecoderPlugin.cxx +++ b/src/decoder/plugins/DsfDecoderPlugin.cxx @@ -39,6 +39,8 @@ #include "tag/TagHandler.hxx" #include "Log.hxx" +#include + static constexpr unsigned DSF_BLOCK_SIZE = 4096; struct DsfMetaData { @@ -188,6 +190,13 @@ bit_reverse_buffer(uint8_t *p, uint8_t *end) *p = bit_reverse(*p); } +static void +InterleaveDsfBlockMono(uint8_t *gcc_restrict dest, + const uint8_t *gcc_restrict src) +{ + memcpy(dest, src, DSF_BLOCK_SIZE); +} + /** * DSF data is build up of alternating 4096 blocks of DSD samples for left and * right. Convert the buffer holding 1 block of 4096 DSD left samples and 1 @@ -195,7 +204,8 @@ bit_reverse_buffer(uint8_t *p, uint8_t *end) * order. */ static void -dsf_to_pcm_order(uint8_t *gcc_restrict dest, const uint8_t *gcc_restrict src) +InterleaveDsfBlockStereo(uint8_t *gcc_restrict dest, + const uint8_t *gcc_restrict src) { for (size_t i = 0; i < DSF_BLOCK_SIZE; ++i) { dest[2 * i] = src[i]; @@ -203,6 +213,36 @@ dsf_to_pcm_order(uint8_t *gcc_restrict dest, const uint8_t *gcc_restrict src) } } +static void +InterleaveDsfBlockChannel(uint8_t *gcc_restrict dest, + const uint8_t *gcc_restrict src, + unsigned channels) +{ + for (size_t i = 0; i < DSF_BLOCK_SIZE; ++i, dest += channels, ++src) + *dest = *src; +} + +static void +InterleaveDsfBlockGeneric(uint8_t *gcc_restrict dest, + const uint8_t *gcc_restrict src, + unsigned channels) +{ + for (unsigned c = 0; c < channels; ++c, ++dest, src += DSF_BLOCK_SIZE) + InterleaveDsfBlockChannel(dest, src, channels); +} + +static void +InterleaveDsfBlock(uint8_t *gcc_restrict dest, const uint8_t *gcc_restrict src, + unsigned channels) +{ + if (channels == 1) + InterleaveDsfBlockMono(dest, src); + else if (channels == 2) + InterleaveDsfBlockStereo(dest, src); + else + InterleaveDsfBlockGeneric(dest, src, channels); +} + /** * Decode one complete DSF 'data' chunk i.e. a complete song */ @@ -212,14 +252,10 @@ dsf_decode_chunk(Decoder &decoder, InputStream &is, offset_type chunk_size, bool bitreverse) { - uint8_t buffer[DSF_BLOCK_SIZE * 2]; + /* worst-case buffer size */ + uint8_t buffer[MAX_CHANNELS * DSF_BLOCK_SIZE]; - const size_t sample_size = sizeof(buffer[0]); - const size_t frame_size = channels * sample_size; - const unsigned buffer_frames = sizeof(buffer) / frame_size; - const unsigned buffer_samples = buffer_frames * frame_size; - const size_t buffer_size = buffer_samples * sample_size; - const size_t block_size = buffer_size; + const size_t block_size = channels * DSF_BLOCK_SIZE; while (chunk_size >= block_size) { chunk_size -= block_size; @@ -230,8 +266,8 @@ dsf_decode_chunk(Decoder &decoder, InputStream &is, if (bitreverse) bit_reverse_buffer(buffer, buffer + block_size); - uint8_t interleaved_buffer[DSF_BLOCK_SIZE * 2]; - dsf_to_pcm_order(interleaved_buffer, buffer); + uint8_t interleaved_buffer[MAX_CHANNELS * DSF_BLOCK_SIZE]; + InterleaveDsfBlock(interleaved_buffer, buffer, channels); const auto cmd = decoder_data(decoder, is, interleaved_buffer, block_size, -- cgit v1.2.3