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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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31
src/encoder/FlacEncoderPlugin.cxx
View File

@@ -20,7 +20,7 @@
#include "config.h"
#include "FlacEncoderPlugin.hxx"
#include "EncoderAPI.hxx"
#include "audio_format.h"
#include "AudioFormat.hxx"
#include "pcm/PcmBuffer.hxx"
#include "util/fifo_buffer.h"
@@ -40,7 +40,7 @@ extern "C" {
struct flac_encoder {
Encoder encoder;
struct audio_format audio_format;
AudioFormat audio_format;
unsigned compression;
FLAC__StreamEncoder *fse;
@@ -160,31 +160,31 @@ flac_encoder_close(Encoder *_encoder)
}
static bool
flac_encoder_open(Encoder *_encoder, struct audio_format *audio_format,
flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format,
GError **error)
{
struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
unsigned bits_per_sample;
encoder->audio_format = *audio_format;
encoder->audio_format = audio_format;
/* FIXME: flac should support 32bit as well */
switch (audio_format->format) {
case SAMPLE_FORMAT_S8:
switch (audio_format.format) {
case SampleFormat::S8:
bits_per_sample = 8;
break;
case SAMPLE_FORMAT_S16:
case SampleFormat::S16:
bits_per_sample = 16;
break;
case SAMPLE_FORMAT_S24_P32:
case SampleFormat::S24_P32:
bits_per_sample = 24;
break;
default:
bits_per_sample = 24;
audio_format->format = SAMPLE_FORMAT_S24_P32;
audio_format.format = SampleFormat::S24_P32;
}
/* allocate the encoder */
@@ -263,30 +263,33 @@ flac_encoder_write(Encoder *_encoder,
/* format conversion */
num_frames = length / audio_format_frame_size(&encoder->audio_format);
num_frames = length / encoder->audio_format.GetFrameSize();
num_samples = num_frames * encoder->audio_format.channels;
switch (encoder->audio_format.format) {
case SAMPLE_FORMAT_S8:
case SampleFormat::S8:
exbuffer = encoder->expand_buffer.Get(length * 4);
pcm8_to_flac((int32_t *)exbuffer, (const int8_t *)data,
num_samples);
buffer = exbuffer;
break;
case SAMPLE_FORMAT_S16:
case SampleFormat::S16:
exbuffer = encoder->expand_buffer.Get(length * 2);
pcm16_to_flac((int32_t *)exbuffer, (const int16_t *)data,
num_samples);
buffer = exbuffer;
break;
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
case SampleFormat::S24_P32:
case SampleFormat::S32:
/* nothing need to be done; format is the same for
both mpd and libFLAC */
buffer = data;
break;
default:
gcc_unreachable();
}
/* feed samples to encoder */

14
src/encoder/LameEncoderPlugin.cxx
View File

@@ -20,7 +20,7 @@
#include "config.h"
#include "LameEncoderPlugin.hxx"
#include "EncoderAPI.hxx"
#include "audio_format.h"
#include "AudioFormat.hxx"
#include <lame/lame.h>
@@ -32,7 +32,7 @@
struct LameEncoder final {
Encoder encoder;
struct audio_format audio_format;
AudioFormat audio_format;
float quality;
int bitrate;
@@ -187,15 +187,15 @@ lame_encoder_setup(LameEncoder *encoder, GError **error)
}
static bool
lame_encoder_open(Encoder *_encoder, struct audio_format *audio_format,
lame_encoder_open(Encoder *_encoder, AudioFormat &audio_format,
GError **error)
{
LameEncoder *encoder = (LameEncoder *)_encoder;
audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
audio_format.format = SampleFormat::S16;
audio_format.channels = 2;
encoder->audio_format = *audio_format;
encoder->audio_format = audio_format;
encoder->gfp = lame_init();
if (encoder->gfp == nullptr) {
@@ -233,7 +233,7 @@ lame_encoder_write(Encoder *_encoder,
assert(encoder->buffer_length == 0);
const unsigned num_frames =
length / audio_format_frame_size(&encoder->audio_format);
length / encoder->audio_format.GetFrameSize();
float *left = g_new(float, num_frames);
float *right = g_new(float, num_frames);

2
src/encoder/NullEncoderPlugin.cxx
View File

@@ -66,7 +66,7 @@ null_encoder_close(Encoder *_encoder)
static bool
null_encoder_open(Encoder *_encoder,
gcc_unused struct audio_format *audio_format,
gcc_unused AudioFormat &audio_format,
gcc_unused GError **error)
{
NullEncoder *encoder = (NullEncoder *)_encoder;

36
src/encoder/OpusEncoderPlugin.cxx
View File

@@ -21,7 +21,7 @@
#include "OpusEncoderPlugin.hxx"
#include "OggStream.hxx"
#include "EncoderAPI.hxx"
#include "audio_format.h"
#include "AudioFormat.hxx"
#include "mpd_error.h"
#include <opus.h>
@@ -44,7 +44,7 @@ struct opus_encoder {
/* runtime information */
struct audio_format audio_format;
AudioFormat audio_format;
size_t frame_size;
@@ -144,37 +144,37 @@ opus_encoder_finish(Encoder *_encoder)
static bool
opus_encoder_open(Encoder *_encoder,
struct audio_format *audio_format,
AudioFormat &audio_format,
GError **error_r)
{
struct opus_encoder *encoder = (struct opus_encoder *)_encoder;
/* libopus supports only 48 kHz */
audio_format->sample_rate = 48000;
audio_format.sample_rate = 48000;
if (audio_format->channels > 2)
audio_format->channels = 1;
if (audio_format.channels > 2)
audio_format.channels = 1;
switch ((enum sample_format)audio_format->format) {
case SAMPLE_FORMAT_S16:
case SAMPLE_FORMAT_FLOAT:
switch (audio_format.format) {
case SampleFormat::S16:
case SampleFormat::FLOAT:
break;
case SAMPLE_FORMAT_S8:
audio_format->format = SAMPLE_FORMAT_S16;
case SampleFormat::S8:
audio_format.format = SampleFormat::S16;
break;
default:
audio_format->format = SAMPLE_FORMAT_FLOAT;
audio_format.format = SampleFormat::FLOAT;
break;
}
encoder->audio_format = *audio_format;
encoder->frame_size = audio_format_frame_size(audio_format);
encoder->audio_format = audio_format;
encoder->frame_size = audio_format.GetFrameSize();
int error;
encoder->enc = opus_encoder_create(audio_format->sample_rate,
audio_format->channels,
encoder->enc = opus_encoder_create(audio_format.sample_rate,
audio_format.channels,
OPUS_APPLICATION_AUDIO,
&error);
if (encoder->enc == nullptr) {
@@ -190,7 +190,7 @@ opus_encoder_open(Encoder *_encoder,
opus_encoder_ctl(encoder->enc, OPUS_GET_LOOKAHEAD(&encoder->lookahead));
encoder->buffer_frames = audio_format->sample_rate / 50;
encoder->buffer_frames = audio_format.sample_rate / 50;
encoder->buffer_size = encoder->frame_size * encoder->buffer_frames;
encoder->buffer_position = 0;
encoder->buffer = (unsigned char *)g_malloc(encoder->buffer_size);
@@ -218,7 +218,7 @@ opus_encoder_do_encode(struct opus_encoder *encoder, bool eos,
assert(encoder->buffer_position == encoder->buffer_size);
opus_int32 result =
encoder->audio_format.format == SAMPLE_FORMAT_S16
encoder->audio_format.format == SampleFormat::S16
? opus_encode(encoder->enc,
(const opus_int16 *)encoder->buffer,
encoder->buffer_frames,

14
src/encoder/TwolameEncoderPlugin.cxx
View File

@@ -20,7 +20,7 @@
#include "config.h"
#include "TwolameEncoderPlugin.hxx"
#include "EncoderAPI.hxx"
#include "audio_format.h"
#include "AudioFormat.hxx"
#include <twolame.h>
@@ -32,7 +32,7 @@
struct TwolameEncoder final {
Encoder encoder;
struct audio_format audio_format;
AudioFormat audio_format;
float quality;
int bitrate;
@@ -187,15 +187,15 @@ twolame_encoder_setup(TwolameEncoder *encoder, GError **error)
}
static bool
twolame_encoder_open(Encoder *_encoder, struct audio_format *audio_format,
twolame_encoder_open(Encoder *_encoder, AudioFormat &audio_format,
GError **error)
{
TwolameEncoder *encoder = (TwolameEncoder *)_encoder;
audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
audio_format.format = SampleFormat::S16;
audio_format.channels = 2;
encoder->audio_format = *audio_format;
encoder->audio_format = audio_format;
encoder->options = twolame_init();
if (encoder->options == nullptr) {
@@ -243,7 +243,7 @@ twolame_encoder_write(Encoder *_encoder,
assert(encoder->buffer_length == 0);
const unsigned num_frames =
length / audio_format_frame_size(&encoder->audio_format);
length / encoder->audio_format.GetFrameSize();
int bytes_out = twolame_encode_buffer_interleaved(encoder->options,
src, num_frames,

13
src/encoder/VorbisEncoderPlugin.cxx
View File

@@ -22,7 +22,7 @@
#include "OggStream.hxx"
#include "EncoderAPI.hxx"
#include "Tag.hxx"
#include "audio_format.h"
#include "AudioFormat.hxx"
#include "mpd_error.h"
#include <vorbis/vorbisenc.h>
@@ -43,7 +43,7 @@ struct vorbis_encoder {
/* runtime information */
struct audio_format audio_format;
AudioFormat audio_format;
vorbis_dsp_state vd;
vorbis_block vb;
@@ -202,14 +202,14 @@ vorbis_encoder_send_header(struct vorbis_encoder *encoder)
static bool
vorbis_encoder_open(Encoder *_encoder,
struct audio_format *audio_format,
AudioFormat &audio_format,
GError **error)
{
struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
audio_format->format = SAMPLE_FORMAT_FLOAT;
audio_format.format = SampleFormat::FLOAT;
encoder->audio_format = *audio_format;
encoder->audio_format = audio_format;
if (!vorbis_encoder_reinit(encoder, error))
return false;
@@ -328,8 +328,7 @@ vorbis_encoder_write(Encoder *_encoder,
{
struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
unsigned num_frames = length
/ audio_format_frame_size(&encoder->audio_format);
unsigned num_frames = length / encoder->audio_format.GetFrameSize();
/* this is for only 16-bit audio */

22
src/encoder/WaveEncoderPlugin.cxx
View File

@@ -100,32 +100,32 @@ wave_encoder_finish(Encoder *_encoder)
static bool
wave_encoder_open(Encoder *_encoder,
gcc_unused struct audio_format *audio_format,
AudioFormat &audio_format,
gcc_unused GError **error)
{
WaveEncoder *encoder = (WaveEncoder *)_encoder;
assert(audio_format_valid(audio_format));
assert(audio_format.IsValid());
switch (audio_format->format) {
case SAMPLE_FORMAT_S8:
switch (audio_format.format) {
case SampleFormat::S8:
encoder->bits = 8;
break;
case SAMPLE_FORMAT_S16:
case SampleFormat::S16:
encoder->bits = 16;
break;
case SAMPLE_FORMAT_S24_P32:
case SampleFormat::S24_P32:
encoder->bits = 24;
break;
case SAMPLE_FORMAT_S32:
case SampleFormat::S32:
encoder->bits = 32;
break;
default:
audio_format->format = SAMPLE_FORMAT_S16;
audio_format.format = SampleFormat::S16;
encoder->bits = 16;
break;
}
@@ -136,10 +136,10 @@ wave_encoder_open(Encoder *_encoder,
/* create PCM wave header in initial buffer */
fill_wave_header(header,
audio_format->channels,
audio_format.channels,
encoder->bits,
audio_format->sample_rate,
(encoder->bits / 8) * audio_format->channels );
audio_format.sample_rate,
(encoder->bits / 8) * audio_format.channels);
fifo_buffer_append(encoder->buffer, sizeof(*header));
return true;