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audio_format: convert to C++

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This commit is contained in:
Max Kellermann
2013-08-03 21:00:50 +02:00
parent 67f591a9ce
commit d1e7b4e381
121 changed files with 1251 additions and 1275 deletions
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7
src/decoder/AdPlugDecoderPlugin.cxx
View File

@@ -60,11 +60,10 @@ adplug_file_decode(struct decoder *decoder, const char *path_fs)
if (player == nullptr)
return;
struct audio_format audio_format;
audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2);
assert(audio_format_valid(&audio_format));
const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2);
assert(audio_format.IsValid());
decoder_initialized(decoder, &audio_format, false,
decoder_initialized(decoder, audio_format, false,
player->songlength() / 1000.);
int16_t buffer[2048];

20
src/decoder/AudiofileDecoderPlugin.cxx
View File

@@ -114,27 +114,27 @@ setup_virtual_fops(struct input_stream *stream)
return vf;
}
static enum sample_format
static SampleFormat
audiofile_bits_to_sample_format(int bits)
{
switch (bits) {
case 8:
return SAMPLE_FORMAT_S8;
return SampleFormat::S8;
case 16:
return SAMPLE_FORMAT_S16;
return SampleFormat::S16;
case 24:
return SAMPLE_FORMAT_S24_P32;
return SampleFormat::S24_P32;
case 32:
return SAMPLE_FORMAT_S32;
return SampleFormat::S32;
}
return SAMPLE_FORMAT_UNDEFINED;
return SampleFormat::UNDEFINED;
}
static enum sample_format
static SampleFormat
audiofile_setup_sample_format(AFfilehandle af_fp)
{
int fs, bits;
@@ -160,7 +160,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
struct audio_format audio_format;
AudioFormat audio_format;
float total_time;
uint16_t bit_rate;
int ret;
@@ -180,7 +180,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
return;
}
if (!audio_format_init_checked(&audio_format,
if (!audio_format_init_checked(audio_format,
afGetRate(af_fp, AF_DEFAULT_TRACK),
audiofile_setup_sample_format(af_fp),
afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
@@ -199,7 +199,7 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
decoder_initialized(decoder, audio_format, true, total_time);
do {
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,

14
src/decoder/DsdiffDecoderPlugin.cxx
View File

@@ -433,9 +433,9 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
return;
GError *error = nullptr;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
SampleFormat::DSD,
metadata.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
@@ -448,7 +448,7 @@ dsdiff_stream_decode(struct decoder *decoder, struct input_stream *is)
(float) metadata.sample_rate;
/* success: file was recognized */
decoder_initialized(decoder, &audio_format, false, songtime);
decoder_initialized(decoder, audio_format, false, songtime);
/* every iteration of the following loop decodes one "DSD"
chunk from a DFF file */
@@ -487,9 +487,9 @@ dsdiff_scan_stream(struct input_stream *is,
if (!dsdiff_read_metadata(nullptr, is, &metadata, &chunk_header))
return false;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
SampleFormat::DSD,
metadata.channels, nullptr))
/* refuse to parse files which we cannot play anyway */
return false;

14
src/decoder/DsfDecoderPlugin.cxx
View File

@@ -285,9 +285,9 @@ dsf_stream_decode(struct decoder *decoder, struct input_stream *is)
return;
GError *error = NULL;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
SampleFormat::DSD,
metadata.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
@@ -299,7 +299,7 @@ dsf_stream_decode(struct decoder *decoder, struct input_stream *is)
(float) metadata.sample_rate;
/* success: file was recognized */
decoder_initialized(decoder, &audio_format, false, songtime);
decoder_initialized(decoder, audio_format, false, songtime);
if (!dsf_decode_chunk(decoder, is, metadata.channels,
chunk_size,
@@ -317,9 +317,9 @@ dsf_scan_stream(struct input_stream *is,
if (!dsf_read_metadata(NULL, is, &metadata))
return false;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
SAMPLE_FORMAT_DSD,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
SampleFormat::DSD,
metadata.channels, NULL))
/* refuse to parse files which we cannot play anyway */
return false;

14
src/decoder/FaadDecoderPlugin.cxx
View File

@@ -248,7 +248,7 @@ faad_song_duration(DecoderBuffer *buffer, struct input_stream *is)
*/
static bool
faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
struct audio_format *audio_format, GError **error_r)
AudioFormat &audio_format, GError **error_r)
{
int32_t nbytes;
uint32_t sample_rate;
@@ -285,7 +285,7 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
decoder_buffer_consume(buffer, nbytes);
return audio_format_init_checked(audio_format, sample_rate,
SAMPLE_FORMAT_S16, channels, error_r);
SampleFormat::S16, channels, error_r);
}
/**
@@ -325,7 +325,7 @@ faad_get_file_time_float(struct input_stream *is)
if (length < 0) {
bool ret;
struct audio_format audio_format;
AudioFormat audio_format;
NeAACDecHandle decoder = NeAACDecOpen();
@@ -336,7 +336,7 @@ faad_get_file_time_float(struct input_stream *is)
decoder_buffer_fill(buffer);
ret = faad_decoder_init(decoder, buffer, &audio_format, nullptr);
ret = faad_decoder_init(decoder, buffer, audio_format, nullptr);
if (ret)
length = 0;
@@ -370,7 +370,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
{
GError *error = nullptr;
float total_time = 0;
struct audio_format audio_format;
AudioFormat audio_format;
bool ret;
uint16_t bit_rate = 0;
DecoderBuffer *buffer;
@@ -400,7 +400,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
/* initialize it */
ret = faad_decoder_init(decoder, buffer, &audio_format, &error);
ret = faad_decoder_init(decoder, buffer, audio_format, &error);
if (!ret) {
g_warning("%s", error->message);
g_error_free(error);
@@ -410,7 +410,7 @@ faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
/* initialize the MPD core */
decoder_initialized(mpd_decoder, &audio_format, false, total_time);
decoder_initialized(mpd_decoder, audio_format, false, total_time);
/* the decoder loop */

25
src/decoder/FfmpegDecoderPlugin.cxx
View File

@@ -52,6 +52,11 @@ extern "C" {
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "ffmpeg"
/* suppress the ffmpeg compatibility macro */
#ifdef SampleFormat
#undef SampleFormat
#endif
static GLogLevelFlags
level_ffmpeg_to_glib(int level)
{
@@ -297,20 +302,20 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
}
G_GNUC_CONST
static enum sample_format
static SampleFormat
ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
{
switch (sample_fmt) {
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P:
return SAMPLE_FORMAT_S16;
return SampleFormat::S16;
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_S32P:
return SAMPLE_FORMAT_S32;
return SampleFormat::S32;
case AV_SAMPLE_FMT_FLTP:
return SAMPLE_FORMAT_FLOAT;
return SampleFormat::FLOAT;
default:
break;
@@ -325,7 +330,7 @@ ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
else
g_warning("Unsupported libavcodec SampleFormat value: %d",
sample_fmt);
return SAMPLE_FORMAT_UNDEFINED;
return SampleFormat::UNDEFINED;
}
static AVInputFormat *
@@ -420,14 +425,14 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
return;
}
const enum sample_format sample_format =
const SampleFormat sample_format =
ffmpeg_sample_format(codec_context->sample_fmt);
if (sample_format == SAMPLE_FORMAT_UNDEFINED)
if (sample_format == SampleFormat::UNDEFINED)
return;
GError *error = NULL;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format,
codec_context->sample_rate,
sample_format,
codec_context->channels, &error)) {
@@ -455,7 +460,7 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
? format_context->duration / AV_TIME_BASE
: 0;
decoder_initialized(decoder, &audio_format,
decoder_initialized(decoder, audio_format,
input->seekable, total_time);
AVFrame *frame = avcodec_alloc_frame();

5
src/decoder/FfmpegMetaData.hxx
View File

@@ -26,6 +26,11 @@ extern "C" {
#include <libavutil/dict.h>
}
/* suppress the ffmpeg compatibility macro */
#ifdef SampleFormat
#undef SampleFormat
#endif
struct tag_handler;
void

24
src/decoder/FlacCommon.cxx
View File

@@ -40,24 +40,24 @@ flac_data::flac_data(struct decoder *_decoder,
{
}
static enum sample_format
static SampleFormat
flac_sample_format(unsigned bits_per_sample)
{
switch (bits_per_sample) {
case 8:
return SAMPLE_FORMAT_S8;
return SampleFormat::S8;
case 16:
return SAMPLE_FORMAT_S16;
return SampleFormat::S16;
case 24:
return SAMPLE_FORMAT_S24_P32;
return SampleFormat::S24_P32;
case 32:
return SAMPLE_FORMAT_S32;
return SampleFormat::S32;
default:
return SAMPLE_FORMAT_UNDEFINED;
return SampleFormat::UNDEFINED;
}
}
@@ -69,7 +69,7 @@ flac_got_stream_info(struct flac_data *data,
return;
GError *error = nullptr;
if (!audio_format_init_checked(&data->audio_format,
if (!audio_format_init_checked(data->audio_format,
stream_info->sample_rate,
flac_sample_format(stream_info->bits_per_sample),
stream_info->channels, &error)) {
@@ -79,7 +79,7 @@ flac_got_stream_info(struct flac_data *data,
return;
}
data->frame_size = audio_format_frame_size(&data->audio_format);
data->frame_size = data->audio_format.GetFrameSize();
if (data->total_frames == 0)
data->total_frames = stream_info->total_samples;
@@ -132,7 +132,7 @@ flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header)
return false;
GError *error = nullptr;
if (!audio_format_init_checked(&data->audio_format,
if (!audio_format_init_checked(data->audio_format,
header->sample_rate,
flac_sample_format(header->bits_per_sample),
header->channels, &error)) {
@@ -142,9 +142,9 @@ flac_got_first_frame(struct flac_data *data, const FLAC__FrameHeader *header)
return false;
}
data->frame_size = audio_format_frame_size(&data->audio_format);
data->frame_size = data->audio_format.GetFrameSize();
decoder_initialized(data->decoder, &data->audio_format,
decoder_initialized(data->decoder, data->audio_format,
data->input_stream->seekable,
(float)data->total_frames /
(float)data->audio_format.sample_rate);
@@ -170,7 +170,7 @@ flac_common_write(struct flac_data *data, const FLAC__Frame * frame,
buffer = data->buffer.Get(buffer_size);
flac_convert(buffer, frame->header.channels,
(enum sample_format)data->audio_format.format, buf,
data->audio_format.format, buf,
0, frame->header.blocksize);
if (nbytes > 0)

2
src/decoder/FlacCommon.hxx
View File

@@ -56,7 +56,7 @@ struct flac_data : public FlacInput {
* The validated audio format of the FLAC file. This
* attribute is defined if "initialized" is true.
*/
struct audio_format audio_format;
AudioFormat audio_format;
/**
* The total number of frames in this song. The decoder

2
src/decoder/FlacDecoderPlugin.cxx
View File

@@ -144,7 +144,7 @@ flac_decoder_initialize(struct flac_data *data, FLAC__StreamDecoder *sd,
if (data->initialized) {
/* done */
decoder_initialized(data->decoder, &data->audio_format,
decoder_initialized(data->decoder, data->audio_format,
data->input_stream->seekable,
(float)data->total_frames /
(float)data->audio_format.sample_rate);

16
src/decoder/FlacPcm.cxx
View File

@@ -76,12 +76,12 @@ flac_convert_8(int8_t *dest,
void
flac_convert(void *dest,
unsigned int num_channels, enum sample_format sample_format,
unsigned int num_channels, SampleFormat sample_format,
const FLAC__int32 *const buf[],
unsigned int position, unsigned int end)
{
switch (sample_format) {
case SAMPLE_FORMAT_S16:
case SampleFormat::S16:
if (num_channels == 2)
flac_convert_stereo16((int16_t*)dest, buf,
position, end);
@@ -90,20 +90,20 @@ flac_convert(void *dest,
position, end);
break;
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
case SampleFormat::S24_P32:
case SampleFormat::S32:
flac_convert_32((int32_t*)dest, num_channels, buf,
position, end);
break;
case SAMPLE_FORMAT_S8:
case SampleFormat::S8:
flac_convert_8((int8_t*)dest, num_channels, buf,
position, end);
break;
case SAMPLE_FORMAT_FLOAT:
case SAMPLE_FORMAT_DSD:
case SAMPLE_FORMAT_UNDEFINED:
case SampleFormat::FLOAT:
case SampleFormat::DSD:
case SampleFormat::UNDEFINED:
assert(false);
gcc_unreachable();
}

4
src/decoder/FlacPcm.hxx
View File

@@ -20,13 +20,13 @@
#ifndef MPD_FLAC_PCM_HXX
#define MPD_FLAC_PCM_HXX
#include "audio_format.h"
#include "AudioFormat.hxx"
#include <FLAC/ordinals.h>
void
flac_convert(void *dest,
unsigned int num_channels, enum sample_format sample_format,
unsigned int num_channels, SampleFormat sample_format,
const FLAC__int32 *const buf[],
unsigned int position, unsigned int end);

5
src/decoder/FluidsynthDecoderPlugin.cxx
View File

@@ -166,9 +166,8 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
/* initialization complete - announce the audio format to the
MPD core */
struct audio_format audio_format;
audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2);
decoder_initialized(decoder, &audio_format, false, -1);
const AudioFormat audio_format(sample_rate, SampleFormat::S16, 2);
decoder_initialized(decoder, audio_format, false, -1);
while (fluid_player_get_status(player) == FLUID_PLAYER_PLAYING) {
int16_t buffer[2048];

8
src/decoder/GmeDecoderPlugin.cxx
View File

@@ -153,9 +153,9 @@ gme_file_decode(struct decoder *decoder, const char *path_fs)
/* initialize the MPD decoder */
GError *error = nullptr;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, GME_SAMPLE_RATE,
SAMPLE_FORMAT_S16, GME_CHANNELS,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, GME_SAMPLE_RATE,
SampleFormat::S16, GME_CHANNELS,
&error)) {
g_warning("%s", error->message);
g_error_free(error);
@@ -164,7 +164,7 @@ gme_file_decode(struct decoder *decoder, const char *path_fs)
return;
}
decoder_initialized(decoder, &audio_format, true, song_len);
decoder_initialized(decoder, audio_format, true, song_len);
gme_err = gme_start_track(emu, song_num);
if (gme_err != nullptr)

8
src/decoder/MadDecoderPlugin.cxx
View File

@@ -1124,11 +1124,11 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
return;
}
struct audio_format audio_format;
AudioFormat audio_format;
GError *error = nullptr;
if (!audio_format_init_checked(&audio_format,
if (!audio_format_init_checked(audio_format,
data.frame.header.samplerate,
SAMPLE_FORMAT_S24_P32,
SampleFormat::S24_P32,
MAD_NCHANNELS(&data.frame.header),
&error)) {
g_warning("%s", error->message);
@@ -1138,7 +1138,7 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
return;
}
decoder_initialized(decoder, &audio_format,
decoder_initialized(decoder, audio_format,
input_stream_is_seekable(input_stream),
data.total_time);

7
src/decoder/MikmodDecoderPlugin.cxx
View File

@@ -147,7 +147,6 @@ mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs)
{
char *path2;
MODULE *handle;
struct audio_format audio_format;
int ret;
SBYTE buffer[MIKMOD_FRAME_SIZE];
enum decoder_command cmd = DECODE_COMMAND_NONE;
@@ -164,10 +163,10 @@ mikmod_decoder_file_decode(struct decoder *decoder, const char *path_fs)
/* Prevent module from looping forever */
handle->loop = 0;
audio_format_init(&audio_format, mikmod_sample_rate, SAMPLE_FORMAT_S16, 2);
assert(audio_format_valid(&audio_format));
const AudioFormat audio_format(mikmod_sample_rate, SampleFormat::S16, 2);
assert(audio_format.IsValid());
decoder_initialized(decoder, &audio_format, false, 0);
decoder_initialized(decoder, audio_format, false, 0);
Player_Start(handle);
while (cmd == DECODE_COMMAND_NONE && Player_Active()) {

7
src/decoder/ModplugDecoderPlugin.cxx
View File

@@ -94,7 +94,6 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
ModPlugFile *f;
ModPlug_Settings settings;
GByteArray *bdatas;
struct audio_format audio_format;
int ret;
char audio_buffer[MODPLUG_FRAME_SIZE];
enum decoder_command cmd = DECODE_COMMAND_NONE;
@@ -122,10 +121,10 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
return;
}
audio_format_init(&audio_format, 44100, SAMPLE_FORMAT_S16, 2);
assert(audio_format_valid(&audio_format));
static constexpr AudioFormat audio_format(44100, SampleFormat::S16, 2);
assert(audio_format.IsValid());
decoder_initialized(decoder, &audio_format,
decoder_initialized(decoder, audio_format,
input_stream_is_seekable(is),
ModPlug_GetLength(f) / 1000.0);

8
src/decoder/MpcdecDecoderPlugin.cxx
View File

@@ -154,9 +154,9 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
mpc_demux_get_info(demux, &info);
GError *error = nullptr;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, info.sample_freq,
SAMPLE_FORMAT_S24_P32,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, info.sample_freq,
SampleFormat::S24_P32,
info.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
@@ -173,7 +173,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
decoder_replay_gain(mpd_decoder, &replay_gain_info);
decoder_initialized(mpd_decoder, &audio_format,
decoder_initialized(mpd_decoder, audio_format,
input_stream_is_seekable(is),
mpc_streaminfo_get_length(&info));

14
src/decoder/Mpg123DecoderPlugin.cxx
View File

@@ -56,7 +56,7 @@ mpd_mpg123_finish(void)
*/
static bool
mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
struct audio_format *audio_format)
AudioFormat &audio_format)
{
GError *gerror = nullptr;
char *path_dup;
@@ -90,7 +90,7 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
return false;
}
if (!audio_format_init_checked(audio_format, rate, SAMPLE_FORMAT_S16,
if (!audio_format_init_checked(audio_format, rate, SampleFormat::S16,
channels, &gerror)) {
g_warning("%s", gerror->message);
g_error_free(gerror);
@@ -103,7 +103,6 @@ mpd_mpg123_open(mpg123_handle *handle, const char *path_fs,
static void
mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs)
{
struct audio_format audio_format;
mpg123_handle *handle;
int error;
off_t num_samples;
@@ -119,7 +118,8 @@ mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs)
return;
}
if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
AudioFormat audio_format;
if (!mpd_mpg123_open(handle, path_fs, audio_format)) {
mpg123_delete(handle);
return;
}
@@ -128,7 +128,7 @@ mpd_mpg123_file_decode(struct decoder *decoder, const char *path_fs)
/* tell MPD core we're ready */
decoder_initialized(decoder, &audio_format, true,
decoder_initialized(decoder, audio_format, true,
(float)num_samples /
(float)audio_format.sample_rate);
@@ -198,7 +198,6 @@ static bool
mpd_mpg123_scan_file(const char *path_fs,
const struct tag_handler *handler, void *handler_ctx)
{
struct audio_format audio_format;
mpg123_handle *handle;
int error;
off_t num_samples;
@@ -210,7 +209,8 @@ mpd_mpg123_scan_file(const char *path_fs,
return false;
}
if (!mpd_mpg123_open(handle, path_fs, &audio_format)) {
AudioFormat audio_format;
if (!mpd_mpg123_open(handle, path_fs, audio_format)) {
mpg123_delete(handle);
return false;
}

9
src/decoder/OpusDecoderPlugin.cxx
View File

@@ -202,11 +202,10 @@ MPDOpusDecoder::HandleBOS(const ogg_packet &packet)
return DECODE_COMMAND_STOP;
}
struct audio_format audio_format;
audio_format_init(&audio_format, opus_sample_rate,
SAMPLE_FORMAT_S16, channels);
decoder_initialized(decoder, &audio_format, false, -1);
frame_size = audio_format_frame_size(&audio_format);
const AudioFormat audio_format(opus_sample_rate,
SampleFormat::S16, channels);
decoder_initialized(decoder, audio_format, false, -1);
frame_size = audio_format.GetFrameSize();
/* allocate an output buffer for 16 bit PCM samples big enough
to hold a quarter second, larger than 120ms required by

8
src/decoder/PcmDecoderPlugin.cxx
View File

@@ -36,9 +36,9 @@ extern "C" {
static void
pcm_stream_decode(struct decoder *decoder, struct input_stream *is)
{
static constexpr struct audio_format audio_format = {
static constexpr AudioFormat audio_format = {
44100,
SAMPLE_FORMAT_S16,
SampleFormat::S16,
2,
};
@@ -49,14 +49,14 @@ pcm_stream_decode(struct decoder *decoder, struct input_stream *is)
GError *error = nullptr;
enum decoder_command cmd;
double time_to_size = audio_format_time_to_size(&audio_format);
const double time_to_size = audio_format.GetTimeToSize();
float total_time = -1;
const goffset size = input_stream_get_size(is);
if (size >= 0)
total_time = size / time_to_size;
decoder_initialized(decoder, &audio_format,
decoder_initialized(decoder, audio_format,
input_stream_is_seekable(is), total_time);
do {

14
src/decoder/SndfileDecoderPlugin.cxx
View File

@@ -99,7 +99,7 @@ static SF_VIRTUAL_IO vio = {
* Converts a frame number to a timestamp (in seconds).
*/
static float
frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
frame_to_time(sf_count_t frame, const AudioFormat *audio_format)
{
return (float)frame / (float)audio_format->sample_rate;
}
@@ -108,7 +108,7 @@ frame_to_time(sf_count_t frame, const struct audio_format *audio_format)
* Converts a timestamp (in seconds) to a frame number.
*/
static sf_count_t
time_to_frame(float t, const struct audio_format *audio_format)
time_to_frame(float t, const AudioFormat *audio_format)
{
return (sf_count_t)(t * audio_format->sample_rate);
}
@@ -119,7 +119,6 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
GError *error = nullptr;
SNDFILE *sf;
SF_INFO info;
struct audio_format audio_format;
size_t frame_size;
sf_count_t read_frames, num_frames;
int buffer[4096];
@@ -136,18 +135,19 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
/* for now, always read 32 bit samples. Later, we could lower
MPD's CPU usage by reading 16 bit samples with
sf_readf_short() on low-quality source files. */
if (!audio_format_init_checked(&audio_format, info.samplerate,
SAMPLE_FORMAT_S32,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, info.samplerate,
SampleFormat::S32,
info.channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return;
}
decoder_initialized(decoder, &audio_format, info.seekable,
decoder_initialized(decoder, audio_format, info.seekable,
frame_to_time(info.frames, &audio_format));
frame_size = audio_format_frame_size(&audio_format);
frame_size = audio_format.GetFrameSize();
read_frames = sizeof(buffer) / frame_size;
do {

10
src/decoder/VorbisDecoderPlugin.cxx
View File

@@ -202,12 +202,12 @@ vorbis_stream_decode(struct decoder *decoder,
return;
}
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format, vi->rate,
AudioFormat audio_format;
if (!audio_format_init_checked(audio_format, vi->rate,
#ifdef HAVE_TREMOR
SAMPLE_FORMAT_S16,
SampleFormat::S16,
#else
SAMPLE_FORMAT_FLOAT,
SampleFormat::FLOAT,
#endif
vi->channels, &error)) {
g_warning("%s", error->message);
@@ -219,7 +219,7 @@ vorbis_stream_decode(struct decoder *decoder,
if (total_time < 0)
total_time = 0;
decoder_initialized(decoder, &audio_format, vis.seekable, total_time);
decoder_initialized(decoder, audio_format, vis.seekable, total_time);
enum decoder_command cmd = decoder_get_command(decoder);

24
src/decoder/WavpackDecoderPlugin.cxx
View File

@@ -106,27 +106,27 @@ format_samples_float(G_GNUC_UNUSED int bytes_per_sample, void *buffer,
/**
* Choose a MPD sample format from libwavpacks' number of bits.
*/
static enum sample_format
static SampleFormat
wavpack_bits_to_sample_format(bool is_float, int bytes_per_sample)
{
if (is_float)
return SAMPLE_FORMAT_FLOAT;
return SampleFormat::FLOAT;
switch (bytes_per_sample) {
case 1:
return SAMPLE_FORMAT_S8;
return SampleFormat::S8;
case 2:
return SAMPLE_FORMAT_S16;
return SampleFormat::S16;
case 3:
return SAMPLE_FORMAT_S24_P32;
return SampleFormat::S24_P32;
case 4:
return SAMPLE_FORMAT_S32;
return SampleFormat::S32;
default:
return SAMPLE_FORMAT_UNDEFINED;
return SampleFormat::UNDEFINED;
}
}
@@ -139,8 +139,8 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
{
GError *error = NULL;
bool is_float;
enum sample_format sample_format;
struct audio_format audio_format;
SampleFormat sample_format;
AudioFormat audio_format;
format_samples_t format_samples;
float total_time;
int bytes_per_sample, output_sample_size;
@@ -150,7 +150,7 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
wavpack_bits_to_sample_format(is_float,
WavpackGetBytesPerSample(wpc));
if (!audio_format_init_checked(&audio_format,
if (!audio_format_init_checked(audio_format,
WavpackGetSampleRate(wpc),
sample_format,
WavpackGetNumChannels(wpc), &error)) {
@@ -168,14 +168,14 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
total_time = WavpackGetNumSamples(wpc);
total_time /= audio_format.sample_rate;
bytes_per_sample = WavpackGetBytesPerSample(wpc);
output_sample_size = audio_format_frame_size(&audio_format);
output_sample_size = audio_format.GetFrameSize();
/* wavpack gives us all kind of samples in a 32-bit space */
int32_t chunk[1024];
const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
audio_format.channels;
decoder_initialized(decoder, &audio_format, can_seek, total_time);
decoder_initialized(decoder, audio_format, can_seek, total_time);
enum decoder_command cmd = decoder_get_command(decoder);
while (cmd != DECODE_COMMAND_STOP) {

6
src/decoder/WildmidiDecoderPlugin.cxx
View File

@@ -60,9 +60,9 @@ wildmidi_finish(void)
static void
wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
{
static const struct audio_format audio_format = {
static constexpr AudioFormat audio_format = {
WILDMIDI_SAMPLE_RATE,
SAMPLE_FORMAT_S16,
SampleFormat::S16,
2,
};
midi *wm;
@@ -79,7 +79,7 @@ wildmidi_file_decode(struct decoder *decoder, const char *path_fs)
return;
}
decoder_initialized(decoder, &audio_format, true,
decoder_initialized(decoder, audio_format, true,
info->approx_total_samples / WILDMIDI_SAMPLE_RATE);
do {

7
src/decoder/sidplay_decoder_plugin.cxx
View File

@@ -285,11 +285,10 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
/* initialize the MPD decoder */
struct audio_format audio_format;
audio_format_init(&audio_format, 48000, SAMPLE_FORMAT_S16, channels);
assert(audio_format_valid(&audio_format));
const AudioFormat audio_format(48000, SampleFormat::S16, channels);
assert(audio_format.IsValid());
decoder_initialized(decoder, &audio_format, true, (float)song_len);
decoder_initialized(decoder, audio_format, true, (float)song_len);
/* .. and play */