pcm_export: convert to C++
This commit is contained in:
parent
c654c7630a
commit
1729388634
@ -316,7 +316,7 @@ libevent_a_SOURCES = \
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libpcm_a_SOURCES = \
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src/pcm/pcm_buffer.c src/pcm/pcm_buffer.h \
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src/pcm/pcm_export.c src/pcm/pcm_export.h \
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src/pcm/PcmExport.cxx src/pcm/PcmExport.hxx \
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src/pcm/PcmConvert.cxx src/pcm/PcmConvert.hxx \
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src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h \
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src/pcm/pcm_dsd.c src/pcm/pcm_dsd.h \
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@ -1301,6 +1301,7 @@ test_run_convert_LDADD = \
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$(GLIB_LIBS)
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test_run_output_LDADD = $(MPD_LIBS) \
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$(PCM_LIBS) \
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$(OUTPUT_LIBS) \
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$(ENCODER_LIBS) \
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libmixer_plugins.a \
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@ -21,7 +21,8 @@
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#include "AlsaOutputPlugin.hxx"
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#include "output_api.h"
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#include "MixerList.hxx"
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#include "pcm/pcm_export.h"
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#include "pcm/PcmExport.hxx"
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#include "util/Manual.hxx"
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#include <glib.h>
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#include <alsa/asoundlib.h>
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@ -48,7 +49,7 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
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struct AlsaOutput {
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struct audio_output base;
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struct pcm_export_state pcm_export;
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Manual<PcmExport> pcm_export;
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/**
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* The configured name of the ALSA device; empty for the
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@ -202,7 +203,7 @@ alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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pcm_export_init(&ad->pcm_export);
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ad->pcm_export.Construct();
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return true;
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}
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@ -211,7 +212,7 @@ alsa_output_disable(struct audio_output *ao)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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pcm_export_deinit(&ad->pcm_export);
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ad->pcm_export.Destruct();
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}
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static bool
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@ -659,10 +660,9 @@ alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format,
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if (!success)
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return false;
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pcm_export_open(&ad->pcm_export,
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sample_format(audio_format->format),
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audio_format->channels,
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dsd_usb, shift8, packed, reverse_endian);
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ad->pcm_export->Open(sample_format(audio_format->format),
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audio_format->channels,
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dsd_usb, shift8, packed, reverse_endian);
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return true;
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}
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@ -689,8 +689,7 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
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}
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ad->in_frame_size = audio_format_frame_size(audio_format);
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ad->out_frame_size = pcm_export_frame_size(&ad->pcm_export,
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audio_format);
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ad->out_frame_size = ad->pcm_export->GetFrameSize(*audio_format);
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return true;
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}
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@ -809,7 +808,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
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assert(size % ad->in_frame_size == 0);
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chunk = pcm_export(&ad->pcm_export, chunk, size, &size);
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chunk = ad->pcm_export->Export(chunk, size, size);
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assert(size % ad->out_frame_size == 0);
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@ -822,8 +821,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
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% ad->period_frames;
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size_t bytes_written = ret * ad->out_frame_size;
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return pcm_export_source_size(&ad->pcm_export,
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bytes_written);
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return ad->pcm_export->CalcSourceSize(bytes_written);
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}
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if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
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@ -52,14 +52,15 @@
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#endif
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#ifdef AFMT_S24_PACKED
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#include "pcm/pcm_export.h"
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#include "pcm/PcmExport.hxx"
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#include "util/Manual.hxx"
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#endif
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struct oss_data {
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struct audio_output base;
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#ifdef AFMT_S24_PACKED
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struct pcm_export_state pcm_export;
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Manual<PcmExport> pcm_export;
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#endif
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int fd;
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@ -241,7 +242,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
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{
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struct oss_data *od = (struct oss_data *)ao;
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pcm_export_init(&od->pcm_export);
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od->pcm_export.Construct();
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return true;
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}
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@ -250,7 +251,7 @@ oss_output_disable(struct audio_output *ao)
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{
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struct oss_data *od = (struct oss_data *)ao;
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pcm_export_deinit(&od->pcm_export);
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od->pcm_export.Destruct();
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}
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#endif
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@ -502,7 +503,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
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enum sample_format *sample_format_r,
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int *oss_format_r,
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#ifdef AFMT_S24_PACKED
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struct pcm_export_state *pcm_export,
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PcmExport &pcm_export,
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#endif
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GError **error_r)
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{
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@ -537,7 +538,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
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*oss_format_r = oss_format;
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#ifdef AFMT_S24_PACKED
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pcm_export_open(pcm_export, sample_format, 0, false, false,
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pcm_export.Open(sample_format, 0, false, false,
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oss_format == AFMT_S24_PACKED,
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oss_format == AFMT_S24_PACKED &&
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G_BYTE_ORDER != G_LITTLE_ENDIAN);
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@ -554,7 +555,7 @@ static bool
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oss_setup_sample_format(int fd, struct audio_format *audio_format,
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int *oss_format_r,
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#ifdef AFMT_S24_PACKED
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struct pcm_export_state *pcm_export,
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PcmExport &pcm_export,
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#endif
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GError **error_r)
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{
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@ -633,7 +634,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
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oss_setup_sample_rate(od->fd, audio_format, error_r) &&
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oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
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#ifdef AFMT_S24_PACKED
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&od->pcm_export,
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od->pcm_export,
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#endif
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error_r);
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}
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@ -747,14 +748,14 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
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return 0;
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#ifdef AFMT_S24_PACKED
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chunk = pcm_export(&od->pcm_export, chunk, size, &size);
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chunk = od->pcm_export->Export(chunk, size, size);
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#endif
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while (true) {
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ret = write(od->fd, chunk, size);
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if (ret > 0) {
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#ifdef AFMT_S24_PACKED
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ret = pcm_export_source_size(&od->pcm_export, ret);
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ret = od->pcm_export->CalcSourceSize(ret);
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#endif
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return ret;
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}
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148
src/pcm/PcmExport.cxx
Normal file
148
src/pcm/PcmExport.cxx
Normal file
@ -0,0 +1,148 @@
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/*
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "PcmExport.hxx"
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extern "C" {
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#include "pcm_dsd_usb.h"
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#include "pcm_pack.h"
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#include "util/byte_reverse.h"
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}
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void
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PcmExport::Open(enum sample_format sample_format, unsigned _channels,
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bool _dsd_usb, bool _shift8, bool _pack, bool _reverse_endian)
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{
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assert(audio_valid_sample_format(sample_format));
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assert(!_dsd_usb || audio_valid_channel_count(_channels));
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channels = _channels;
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dsd_usb = _dsd_usb && sample_format == SAMPLE_FORMAT_DSD;
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if (dsd_usb)
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/* after the conversion to DSD-over-USB, the DSD
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samples are stuffed inside fake 24 bit samples */
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sample_format = SAMPLE_FORMAT_S24_P32;
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shift8 = _shift8 && sample_format == SAMPLE_FORMAT_S24_P32;
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pack24 = _pack && sample_format == SAMPLE_FORMAT_S24_P32;
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assert(!shift8 || !pack24);
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reverse_endian = 0;
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if (_reverse_endian) {
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size_t sample_size = pack24
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? 3
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: sample_format_size(sample_format);
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assert(sample_size <= 0xff);
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if (sample_size > 1)
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reverse_endian = sample_size;
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}
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}
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size_t
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PcmExport::GetFrameSize(const struct audio_format &audio_format) const
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{
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if (pack24)
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/* packed 24 bit samples (3 bytes per sample) */
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return audio_format.channels * 3;
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if (dsd_usb)
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/* the DSD-over-USB draft says that DSD 1-bit samples
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are enclosed within 24 bit samples, and MPD's
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representation of 24 bit is padded to 32 bit (4
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bytes per sample) */
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return channels * 4;
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return audio_format_frame_size(&audio_format);
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}
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const void *
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PcmExport::Export(const void *data, size_t size, size_t &dest_size_r)
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{
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if (dsd_usb)
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data = pcm_dsd_to_usb(&dsd_buffer, channels,
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(const uint8_t *)data, size, &size);
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if (pack24) {
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assert(size % 4 == 0);
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const size_t num_samples = size / 4;
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const size_t dest_size = num_samples * 3;
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const uint8_t *src8 = (const uint8_t *)data;
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const uint8_t *src_end8 = src8 + size;
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uint8_t *dest = (uint8_t *)
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pcm_buffer_get(&pack_buffer, dest_size);
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assert(dest != NULL);
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pcm_pack_24(dest, (const int32_t *)src8,
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(const int32_t *)src_end8);
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data = dest;
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size = dest_size;
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} else if (shift8) {
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assert(size % 4 == 0);
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const uint8_t *src8 = (const uint8_t *)data;
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const uint8_t *src_end8 = src8 + size;
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const uint32_t *src = (const uint32_t *)src8;
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const uint32_t *const src_end = (const uint32_t *)src_end8;
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uint32_t *dest = (uint32_t *)
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pcm_buffer_get(&pack_buffer, size);
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data = dest;
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while (src < src_end)
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*dest++ = *src++ << 8;
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}
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if (reverse_endian > 0) {
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assert(reverse_endian >= 2);
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uint8_t *dest = (uint8_t *)
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pcm_buffer_get(&reverse_buffer, size);
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assert(dest != NULL);
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const uint8_t *src = (const uint8_t *)data;
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const uint8_t *src_end = src + size;
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reverse_bytes(dest, src, src_end, reverse_endian);
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data = dest;
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}
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dest_size_r = size;
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return data;
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}
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size_t
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PcmExport::CalcSourceSize(size_t size) const
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{
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if (pack24)
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/* 32 bit to 24 bit conversion (4 to 3 bytes) */
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size = (size / 3) * 4;
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if (dsd_usb)
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/* DSD over USB doubles the transport size */
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size /= 2;
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return size;
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}
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@ -1,5 +1,5 @@
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/*
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* Copyright (C) 2003-2012 The Music Player Daemon Project
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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@ -17,15 +17,13 @@
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef PCM_EXPORT_H
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#define PCM_EXPORT_H
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#ifndef PCM_EXPORT_HXX
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#define PCM_EXPORT_HXX
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#include "check.h"
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#include "pcm_buffer.h"
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#include "audio_format.h"
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#include <stdbool.h>
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struct audio_format;
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/**
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@ -33,7 +31,7 @@ struct audio_format;
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* outside of MPD. It has a few more options to tweak the binary
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* representation which are not supported by the pcm_convert library.
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*/
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struct pcm_export_state {
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struct PcmExport {
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/**
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* The buffer is used to convert DSD samples to the
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* DSD-over-USB format.
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@ -85,71 +83,57 @@ struct pcm_export_state {
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* sample (2 or bigger).
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*/
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uint8_t reverse_endian;
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PcmExport() {
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pcm_buffer_init(&reverse_buffer);
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pcm_buffer_init(&pack_buffer);
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pcm_buffer_init(&dsd_buffer);
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}
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~PcmExport() {
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pcm_buffer_deinit(&reverse_buffer);
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pcm_buffer_deinit(&pack_buffer);
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pcm_buffer_deinit(&dsd_buffer);
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}
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/**
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* Open the #pcm_export_state object.
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*
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* There is no "close" method. This function may be called multiple
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* times to reuse the object, until pcm_export_deinit() is called.
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*
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* This function cannot fail.
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*
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* @param channels the number of channels; ignored unless dsd_usb is set
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*/
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void Open(enum sample_format sample_format, unsigned channels,
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bool dsd_usb, bool shift8, bool pack, bool reverse_endian);
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/**
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* Calculate the size of one output frame.
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*/
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gcc_pure
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size_t GetFrameSize(const struct audio_format &audio_format) const;
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/**
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* Export a PCM buffer.
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*
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* @param state an initialized and open pcm_export_state object
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* @param src the source PCM buffer
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* @param src_size the size of #src in bytes
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* @param dest_size_r returns the number of bytes of the destination buffer
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* @return the destination buffer (may be a pointer to the source buffer)
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*/
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const void *Export(const void *src, size_t src_size,
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size_t &dest_size_r);
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/**
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* Converts the number of consumed bytes from the pcm_export()
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* destination buffer to the according number of bytes from the
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* pcm_export() source buffer.
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*/
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gcc_pure
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size_t CalcSourceSize(size_t dest_size) const;
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};
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#ifdef __cplusplus
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extern "C" {
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#endif
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/**
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* Initialize a #pcm_export_state object.
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*/
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void
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pcm_export_init(struct pcm_export_state *state);
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/**
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* Deinitialize a #pcm_export_state object and free allocated memory.
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*/
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void
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pcm_export_deinit(struct pcm_export_state *state);
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/**
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* Open the #pcm_export_state object.
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*
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* There is no "close" method. This function may be called multiple
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* times to reuse the object, until pcm_export_deinit() is called.
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*
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* This function cannot fail.
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*
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* @param channels the number of channels; ignored unless dsd_usb is set
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*/
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void
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pcm_export_open(struct pcm_export_state *state,
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enum sample_format sample_format, unsigned channels,
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bool dsd_usb, bool shift8, bool pack, bool reverse_endian);
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/**
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* Calculate the size of one output frame.
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*/
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G_GNUC_PURE
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size_t
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pcm_export_frame_size(const struct pcm_export_state *state,
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const struct audio_format *audio_format);
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/**
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* Export a PCM buffer.
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*
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* @param state an initialized and open pcm_export_state object
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* @param src the source PCM buffer
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* @param src_size the size of #src in bytes
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* @param dest_size_r returns the number of bytes of the destination buffer
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* @return the destination buffer (may be a pointer to the source buffer)
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*/
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const void *
|
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pcm_export(struct pcm_export_state *state, const void *src, size_t src_size,
|
||||
size_t *dest_size_r);
|
||||
|
||||
/**
|
||||
* Converts the number of consumed bytes from the pcm_export()
|
||||
* destination buffer to the according number of bytes from the
|
||||
* pcm_export() source buffer.
|
||||
*/
|
||||
G_GNUC_PURE
|
||||
size_t
|
||||
pcm_export_source_size(const struct pcm_export_state *state, size_t dest_size);
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
@ -1,160 +0,0 @@
|
||||
/*
|
||||
* Copyright (C) 2003-2012 The Music Player Daemon Project
|
||||
* http://www.musicpd.org
|
||||
*
|
||||
* This program is free software; you can redistribute it and/or modify
|
||||
* it under the terms of the GNU General Public License as published by
|
||||
* the Free Software Foundation; either version 2 of the License, or
|
||||
* (at your option) any later version.
|
||||
*
|
||||
* This program is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
* GNU General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU General Public License along
|
||||
* with this program; if not, write to the Free Software Foundation, Inc.,
|
||||
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
#include "pcm_export.h"
|
||||
#include "pcm_dsd_usb.h"
|
||||
#include "pcm_pack.h"
|
||||
#include "util/byte_reverse.h"
|
||||
|
||||
void
|
||||
pcm_export_init(struct pcm_export_state *state)
|
||||
{
|
||||
pcm_buffer_init(&state->reverse_buffer);
|
||||
pcm_buffer_init(&state->pack_buffer);
|
||||
pcm_buffer_init(&state->dsd_buffer);
|
||||
}
|
||||
|
||||
void pcm_export_deinit(struct pcm_export_state *state)
|
||||
{
|
||||
pcm_buffer_deinit(&state->reverse_buffer);
|
||||
pcm_buffer_deinit(&state->pack_buffer);
|
||||
pcm_buffer_deinit(&state->dsd_buffer);
|
||||
}
|
||||
|
||||
void
|
||||
pcm_export_open(struct pcm_export_state *state,
|
||||
enum sample_format sample_format, unsigned channels,
|
||||
bool dsd_usb, bool shift8, bool pack, bool reverse_endian)
|
||||
{
|
||||
assert(audio_valid_sample_format(sample_format));
|
||||
assert(!dsd_usb || audio_valid_channel_count(channels));
|
||||
|
||||
state->channels = channels;
|
||||
state->dsd_usb = dsd_usb && sample_format == SAMPLE_FORMAT_DSD;
|
||||
if (state->dsd_usb)
|
||||
/* after the conversion to DSD-over-USB, the DSD
|
||||
samples are stuffed inside fake 24 bit samples */
|
||||
sample_format = SAMPLE_FORMAT_S24_P32;
|
||||
|
||||
state->shift8 = shift8 && sample_format == SAMPLE_FORMAT_S24_P32;
|
||||
state->pack24 = pack && sample_format == SAMPLE_FORMAT_S24_P32;
|
||||
|
||||
assert(!state->shift8 || !state->pack24);
|
||||
|
||||
state->reverse_endian = 0;
|
||||
if (reverse_endian) {
|
||||
size_t sample_size = state->pack24
|
||||
? 3
|
||||
: sample_format_size(sample_format);
|
||||
assert(sample_size <= 0xff);
|
||||
|
||||
if (sample_size > 1)
|
||||
state->reverse_endian = sample_size;
|
||||
}
|
||||
}
|
||||
|
||||
size_t
|
||||
pcm_export_frame_size(const struct pcm_export_state *state,
|
||||
const struct audio_format *audio_format)
|
||||
{
|
||||
assert(state != NULL);
|
||||
assert(audio_format != NULL);
|
||||
|
||||
if (state->pack24)
|
||||
/* packed 24 bit samples (3 bytes per sample) */
|
||||
return audio_format->channels * 3;
|
||||
|
||||
if (state->dsd_usb)
|
||||
/* the DSD-over-USB draft says that DSD 1-bit samples
|
||||
are enclosed within 24 bit samples, and MPD's
|
||||
representation of 24 bit is padded to 32 bit (4
|
||||
bytes per sample) */
|
||||
return audio_format->channels * 4;
|
||||
|
||||
return audio_format_frame_size(audio_format);
|
||||
}
|
||||
|
||||
const void *
|
||||
pcm_export(struct pcm_export_state *state, const void *data, size_t size,
|
||||
size_t *dest_size_r)
|
||||
{
|
||||
if (state->dsd_usb)
|
||||
data = pcm_dsd_to_usb(&state->dsd_buffer, state->channels,
|
||||
data, size, &size);
|
||||
|
||||
if (state->pack24) {
|
||||
assert(size % 4 == 0);
|
||||
|
||||
const size_t num_samples = size / 4;
|
||||
const size_t dest_size = num_samples * 3;
|
||||
|
||||
const uint8_t *src8 = data, *src_end8 = src8 + size;
|
||||
uint8_t *dest = pcm_buffer_get(&state->pack_buffer, dest_size);
|
||||
assert(dest != NULL);
|
||||
|
||||
pcm_pack_24(dest, (const int32_t *)src8,
|
||||
(const int32_t *)src_end8);
|
||||
|
||||
data = dest;
|
||||
size = dest_size;
|
||||
} else if (state->shift8) {
|
||||
assert(size % 4 == 0);
|
||||
|
||||
const uint8_t *src8 = data, *src_end8 = src8 + size;
|
||||
const uint32_t *src = (const uint32_t *)src8;
|
||||
const uint32_t *const src_end = (const uint32_t *)src_end8;
|
||||
|
||||
uint32_t *dest = pcm_buffer_get(&state->pack_buffer, size);
|
||||
data = dest;
|
||||
|
||||
while (src < src_end)
|
||||
*dest++ = *src++ << 8;
|
||||
}
|
||||
|
||||
|
||||
if (state->reverse_endian > 0) {
|
||||
assert(state->reverse_endian >= 2);
|
||||
|
||||
void *dest = pcm_buffer_get(&state->reverse_buffer, size);
|
||||
assert(dest != NULL);
|
||||
|
||||
const uint8_t *src = data, *src_end = src + size;
|
||||
reverse_bytes(dest, src, src_end, state->reverse_endian);
|
||||
|
||||
data = dest;
|
||||
}
|
||||
|
||||
*dest_size_r = size;
|
||||
return data;
|
||||
}
|
||||
|
||||
size_t
|
||||
pcm_export_source_size(const struct pcm_export_state *state, size_t size)
|
||||
{
|
||||
if (state->pack24)
|
||||
/* 32 bit to 24 bit conversion (4 to 3 bytes) */
|
||||
size = (size / 3) * 4;
|
||||
|
||||
if (state->dsd_usb)
|
||||
/* DSD over USB doubles the transport size */
|
||||
size /= 2;
|
||||
|
||||
return size;
|
||||
}
|
Loading…
Reference in New Issue
Block a user