decoder/dsdiff: don't convert to PCM
Move the responsibility for the conversion to the PCM library. This will allow passing the verbatim DSD samples to an output plugin.
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c9c57af5f7
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9c36e71081
@ -481,7 +481,6 @@ libdecoder_plugins_a_SOURCES = \
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src/decoder/pcm_decoder_plugin.c \
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src/decoder/dsdiff_decoder_plugin.c \
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src/decoder/dsdiff_decoder_plugin.h \
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src/dsd2pcm/dsd2pcm.c src/dsd2pcm/dsd2pcm.h \
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src/decoder_buffer.c \
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src/decoder_plugin.c \
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src/decoder_list.c
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@ -28,7 +28,6 @@
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#include "dsdiff_decoder_plugin.h"
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#include "decoder_api.h"
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#include "audio_check.h"
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#include "dsd2pcm/dsd2pcm.h"
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#include <unistd.h>
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#include <stdio.h> /* for SEEK_SET, SEEK_CUR */
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@ -55,12 +54,14 @@ struct dsdiff_metadata {
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unsigned sample_rate, channels;
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};
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static bool lsbitfirst;
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static enum sample_format dsd_sample_format;
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static bool
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dsdiff_init(const struct config_param *param)
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{
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lsbitfirst = config_get_block_bool(param, "lsbitfirst", false);
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dsd_sample_format = config_get_block_bool(param, "lsbitfirst", false)
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? SAMPLE_FORMAT_DSD_LSBFIRST
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: SAMPLE_FORMAT_DSD;
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return true;
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}
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@ -306,7 +307,7 @@ dsdiff_read_metadata(struct decoder *decoder, struct input_stream *is,
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static bool
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dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is,
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unsigned channels,
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dsd2pcm_ctx **dsd2pcm, uint64_t chunk_size)
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uint64_t chunk_size)
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{
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uint8_t buffer[8192];
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const size_t sample_size = sizeof(buffer[0]);
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@ -314,18 +315,15 @@ dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is,
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const unsigned buffer_frames = sizeof(buffer) / frame_size;
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const unsigned buffer_samples = buffer_frames * frame_size;
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const size_t buffer_size = buffer_samples * sample_size;
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float f_buffer[G_N_ELEMENTS(buffer)];
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while (chunk_size > 0) {
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/* see how much aligned data from the remaining chunk
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fits into the local buffer */
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unsigned now_frames = buffer_frames;
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size_t now_size = buffer_size;
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unsigned now_samples = buffer_samples;
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if (chunk_size < (uint64_t)now_size) {
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now_frames = (unsigned)chunk_size / frame_size;
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now_size = now_frames * frame_size;
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now_samples = now_frames * channels;
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}
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size_t nbytes = decoder_read(decoder, is, buffer, now_size);
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@ -334,20 +332,8 @@ dsdiff_decode_chunk(struct decoder *decoder, struct input_stream *is,
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chunk_size -= nbytes;
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/* invoke the dsp2pcm library, once for each
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channel */
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for (unsigned c = 0; c < channels; ++c)
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dsd2pcm_translate(dsd2pcm[c], now_frames,
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buffer + c, channels,
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lsbitfirst, f_buffer + c, channels);
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/* convert to integer and submit to the decoder API */
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enum decoder_command cmd =
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decoder_data(decoder, is, f_buffer,
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now_samples * sizeof(f_buffer[0]),
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0);
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decoder_data(decoder, is, buffer, nbytes, 0);
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switch (cmd) {
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case DECODE_COMMAND_NONE:
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break;
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@ -381,25 +367,13 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
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GError *error = NULL;
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struct audio_format audio_format;
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if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
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SAMPLE_FORMAT_FLOAT,
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dsd_sample_format,
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metadata.channels, &error)) {
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g_warning("%s", error->message);
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g_error_free(error);
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return;
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}
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/* initialize the dsd2pcm library */
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dsd2pcm_ctx *dsd2pcm[MAX_CHANNELS];
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for (unsigned i = 0; i < metadata.channels; ++i) {
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dsd2pcm[i] = dsd2pcm_init();
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if (dsd2pcm[i] == NULL) {
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for (unsigned j = 0; j < i; ++j)
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dsd2pcm_destroy(dsd2pcm[j]);
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return;
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}
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}
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/* success: file was recognized */
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decoder_initialized(decoder, &audio_format, false, -1);
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@ -413,7 +387,7 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
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if (dsdiff_id_equals(&chunk_header.id, "DSD ")) {
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if (!dsdiff_decode_chunk(decoder, is,
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metadata.channels,
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dsd2pcm, chunk_size))
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chunk_size))
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break;
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} else {
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/* ignore other chunks */
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@ -428,9 +402,6 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
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if (!dsdiff_read_chunk_header(decoder, is, &chunk_header))
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break;
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}
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for (unsigned i = 0; i < metadata.channels; ++i)
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dsd2pcm_destroy(dsd2pcm[i]);
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}
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static bool
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@ -449,7 +420,7 @@ dsdiff_scan_stream(struct input_stream *is,
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struct audio_format audio_format;
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if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
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SAMPLE_FORMAT_S24_P32,
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dsd_sample_format,
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metadata.channels, NULL))
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/* refuse to parse files which we cannot play anyway */
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return false;
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