pcm_export: convert to C++

This commit is contained in:
Max Kellermann 2013-04-09 01:24:32 +02:00
parent c654c7630a
commit 1729388634
6 changed files with 227 additions and 255 deletions

View File

@ -316,7 +316,7 @@ libevent_a_SOURCES = \
libpcm_a_SOURCES = \
src/pcm/pcm_buffer.c src/pcm/pcm_buffer.h \
src/pcm/pcm_export.c src/pcm/pcm_export.h \
src/pcm/PcmExport.cxx src/pcm/PcmExport.hxx \
src/pcm/PcmConvert.cxx src/pcm/PcmConvert.hxx \
src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h \
src/pcm/pcm_dsd.c src/pcm/pcm_dsd.h \
@ -1301,6 +1301,7 @@ test_run_convert_LDADD = \
$(GLIB_LIBS)
test_run_output_LDADD = $(MPD_LIBS) \
$(PCM_LIBS) \
$(OUTPUT_LIBS) \
$(ENCODER_LIBS) \
libmixer_plugins.a \

View File

@ -21,7 +21,8 @@
#include "AlsaOutputPlugin.hxx"
#include "output_api.h"
#include "MixerList.hxx"
#include "pcm/pcm_export.h"
#include "pcm/PcmExport.hxx"
#include "util/Manual.hxx"
#include <glib.h>
#include <alsa/asoundlib.h>
@ -48,7 +49,7 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
struct AlsaOutput {
struct audio_output base;
struct pcm_export_state pcm_export;
Manual<PcmExport> pcm_export;
/**
* The configured name of the ALSA device; empty for the
@ -202,7 +203,7 @@ alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
AlsaOutput *ad = (AlsaOutput *)ao;
pcm_export_init(&ad->pcm_export);
ad->pcm_export.Construct();
return true;
}
@ -211,7 +212,7 @@ alsa_output_disable(struct audio_output *ao)
{
AlsaOutput *ad = (AlsaOutput *)ao;
pcm_export_deinit(&ad->pcm_export);
ad->pcm_export.Destruct();
}
static bool
@ -659,10 +660,9 @@ alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format,
if (!success)
return false;
pcm_export_open(&ad->pcm_export,
sample_format(audio_format->format),
audio_format->channels,
dsd_usb, shift8, packed, reverse_endian);
ad->pcm_export->Open(sample_format(audio_format->format),
audio_format->channels,
dsd_usb, shift8, packed, reverse_endian);
return true;
}
@ -689,8 +689,7 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
}
ad->in_frame_size = audio_format_frame_size(audio_format);
ad->out_frame_size = pcm_export_frame_size(&ad->pcm_export,
audio_format);
ad->out_frame_size = ad->pcm_export->GetFrameSize(*audio_format);
return true;
}
@ -809,7 +808,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
assert(size % ad->in_frame_size == 0);
chunk = pcm_export(&ad->pcm_export, chunk, size, &size);
chunk = ad->pcm_export->Export(chunk, size, size);
assert(size % ad->out_frame_size == 0);
@ -822,8 +821,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
% ad->period_frames;
size_t bytes_written = ret * ad->out_frame_size;
return pcm_export_source_size(&ad->pcm_export,
bytes_written);
return ad->pcm_export->CalcSourceSize(bytes_written);
}
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&

View File

@ -52,14 +52,15 @@
#endif
#ifdef AFMT_S24_PACKED
#include "pcm/pcm_export.h"
#include "pcm/PcmExport.hxx"
#include "util/Manual.hxx"
#endif
struct oss_data {
struct audio_output base;
#ifdef AFMT_S24_PACKED
struct pcm_export_state pcm_export;
Manual<PcmExport> pcm_export;
#endif
int fd;
@ -241,7 +242,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
struct oss_data *od = (struct oss_data *)ao;
pcm_export_init(&od->pcm_export);
od->pcm_export.Construct();
return true;
}
@ -250,7 +251,7 @@ oss_output_disable(struct audio_output *ao)
{
struct oss_data *od = (struct oss_data *)ao;
pcm_export_deinit(&od->pcm_export);
od->pcm_export.Destruct();
}
#endif
@ -502,7 +503,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
enum sample_format *sample_format_r,
int *oss_format_r,
#ifdef AFMT_S24_PACKED
struct pcm_export_state *pcm_export,
PcmExport &pcm_export,
#endif
GError **error_r)
{
@ -537,7 +538,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
*oss_format_r = oss_format;
#ifdef AFMT_S24_PACKED
pcm_export_open(pcm_export, sample_format, 0, false, false,
pcm_export.Open(sample_format, 0, false, false,
oss_format == AFMT_S24_PACKED,
oss_format == AFMT_S24_PACKED &&
G_BYTE_ORDER != G_LITTLE_ENDIAN);
@ -554,7 +555,7 @@ static bool
oss_setup_sample_format(int fd, struct audio_format *audio_format,
int *oss_format_r,
#ifdef AFMT_S24_PACKED
struct pcm_export_state *pcm_export,
PcmExport &pcm_export,
#endif
GError **error_r)
{
@ -633,7 +634,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
oss_setup_sample_rate(od->fd, audio_format, error_r) &&
oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
#ifdef AFMT_S24_PACKED
&od->pcm_export,
od->pcm_export,
#endif
error_r);
}
@ -747,14 +748,14 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
return 0;
#ifdef AFMT_S24_PACKED
chunk = pcm_export(&od->pcm_export, chunk, size, &size);
chunk = od->pcm_export->Export(chunk, size, size);
#endif
while (true) {
ret = write(od->fd, chunk, size);
if (ret > 0) {
#ifdef AFMT_S24_PACKED
ret = pcm_export_source_size(&od->pcm_export, ret);
ret = od->pcm_export->CalcSourceSize(ret);
#endif
return ret;
}

148
src/pcm/PcmExport.cxx Normal file
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@ -0,0 +1,148 @@
/*
* Copyright (C) 2003-2013 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 "PcmExport.hxx"
extern "C" {
#include "pcm_dsd_usb.h"
#include "pcm_pack.h"
#include "util/byte_reverse.h"
}
void
PcmExport::Open(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));
channels = _channels;
dsd_usb = _dsd_usb && sample_format == SAMPLE_FORMAT_DSD;
if (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;
shift8 = _shift8 && sample_format == SAMPLE_FORMAT_S24_P32;
pack24 = _pack && sample_format == SAMPLE_FORMAT_S24_P32;
assert(!shift8 || !pack24);
reverse_endian = 0;
if (_reverse_endian) {
size_t sample_size = pack24
? 3
: sample_format_size(sample_format);
assert(sample_size <= 0xff);
if (sample_size > 1)
reverse_endian = sample_size;
}
}
size_t
PcmExport::GetFrameSize(const struct audio_format &audio_format) const
{
if (pack24)
/* packed 24 bit samples (3 bytes per sample) */
return audio_format.channels * 3;
if (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 channels * 4;
return audio_format_frame_size(&audio_format);
}
const void *
PcmExport::Export(const void *data, size_t size, size_t &dest_size_r)
{
if (dsd_usb)
data = pcm_dsd_to_usb(&dsd_buffer, channels,
(const uint8_t *)data, size, &size);
if (pack24) {
assert(size % 4 == 0);
const size_t num_samples = size / 4;
const size_t dest_size = num_samples * 3;
const uint8_t *src8 = (const uint8_t *)data;
const uint8_t *src_end8 = src8 + size;
uint8_t *dest = (uint8_t *)
pcm_buffer_get(&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 (shift8) {
assert(size % 4 == 0);
const uint8_t *src8 = (const uint8_t *)data;
const uint8_t *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 = (uint32_t *)
pcm_buffer_get(&pack_buffer, size);
data = dest;
while (src < src_end)
*dest++ = *src++ << 8;
}
if (reverse_endian > 0) {
assert(reverse_endian >= 2);
uint8_t *dest = (uint8_t *)
pcm_buffer_get(&reverse_buffer, size);
assert(dest != NULL);
const uint8_t *src = (const uint8_t *)data;
const uint8_t *src_end = src + size;
reverse_bytes(dest, src, src_end, reverse_endian);
data = dest;
}
dest_size_r = size;
return data;
}
size_t
PcmExport::CalcSourceSize(size_t size) const
{
if (pack24)
/* 32 bit to 24 bit conversion (4 to 3 bytes) */
size = (size / 3) * 4;
if (dsd_usb)
/* DSD over USB doubles the transport size */
size /= 2;
return size;
}

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@ -1,5 +1,5 @@
/*
* Copyright (C) 2003-2012 The Music Player Daemon Project
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@ -17,15 +17,13 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef PCM_EXPORT_H
#define PCM_EXPORT_H
#ifndef PCM_EXPORT_HXX
#define PCM_EXPORT_HXX
#include "check.h"
#include "pcm_buffer.h"
#include "audio_format.h"
#include <stdbool.h>
struct audio_format;
/**
@ -33,7 +31,7 @@ struct audio_format;
* outside of MPD. It has a few more options to tweak the binary
* representation which are not supported by the pcm_convert library.
*/
struct pcm_export_state {
struct PcmExport {
/**
* The buffer is used to convert DSD samples to the
* DSD-over-USB format.
@ -85,71 +83,57 @@ struct pcm_export_state {
* sample (2 or bigger).
*/
uint8_t reverse_endian;
PcmExport() {
pcm_buffer_init(&reverse_buffer);
pcm_buffer_init(&pack_buffer);
pcm_buffer_init(&dsd_buffer);
}
~PcmExport() {
pcm_buffer_deinit(&reverse_buffer);
pcm_buffer_deinit(&pack_buffer);
pcm_buffer_deinit(&dsd_buffer);
}
/**
* Open the #pcm_export_state object.
*
* There is no "close" method. This function may be called multiple
* times to reuse the object, until pcm_export_deinit() is called.
*
* This function cannot fail.
*
* @param channels the number of channels; ignored unless dsd_usb is set
*/
void Open(enum sample_format sample_format, unsigned channels,
bool dsd_usb, bool shift8, bool pack, bool reverse_endian);
/**
* Calculate the size of one output frame.
*/
gcc_pure
size_t GetFrameSize(const struct audio_format &audio_format) const;
/**
* Export a PCM buffer.
*
* @param state an initialized and open pcm_export_state object
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer (may be a pointer to the source buffer)
*/
const void *Export(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.
*/
gcc_pure
size_t CalcSourceSize(size_t dest_size) const;
};
#ifdef __cplusplus
extern "C" {
#endif
/**
* Initialize a #pcm_export_state object.
*/
void
pcm_export_init(struct pcm_export_state *state);
/**
* Deinitialize a #pcm_export_state object and free allocated memory.
*/
void
pcm_export_deinit(struct pcm_export_state *state);
/**
* Open the #pcm_export_state object.
*
* There is no "close" method. This function may be called multiple
* times to reuse the object, until pcm_export_deinit() is called.
*
* This function cannot fail.
*
* @param channels the number of channels; ignored unless dsd_usb is set
*/
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);
/**
* Calculate the size of one output frame.
*/
G_GNUC_PURE
size_t
pcm_export_frame_size(const struct pcm_export_state *state,
const struct audio_format *audio_format);
/**
* Export a PCM buffer.
*
* @param state an initialized and open pcm_export_state object
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer (may be a pointer to the source buffer)
*/
const void *
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

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@ -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;
}