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authorMax Kellermann <max@duempel.org>2009-11-10 17:11:34 +0100
committerMax Kellermann <max@duempel.org>2009-12-02 22:29:50 +0100
commitc412d6251e9cd3abe735b7622af4003502e54f72 (patch)
tree7344c13f62e4cc788c830c05d21bb7b5b47f5866 /src/encoder
parent68c2cfbb4067b2292e1ff1d4e7716ff370903f84 (diff)
downloadmpd-c412d6251e9cd3abe735b7622af4003502e54f72.tar.gz
mpd-c412d6251e9cd3abe735b7622af4003502e54f72.tar.xz
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audio_format: changed "bits" to "enum sample_format"
This patch prepares support for floating point samples (and probably other formats). It changes the meaning of the "bits" attribute from a bit count to a symbolic value.
Diffstat (limited to 'src/encoder')
-rw-r--r--src/encoder/flac_encoder.c44
-rw-r--r--src/encoder/lame_encoder.c2
-rw-r--r--src/encoder/twolame_encoder.c2
-rw-r--r--src/encoder/vorbis_encoder.c2
-rw-r--r--src/encoder/wave_encoder.c29
5 files changed, 61 insertions, 18 deletions
diff --git a/src/encoder/flac_encoder.c b/src/encoder/flac_encoder.c
index ab7dc0c39..4f80fe568 100644
--- a/src/encoder/flac_encoder.c
+++ b/src/encoder/flac_encoder.c
@@ -89,7 +89,8 @@ flac_encoder_finish(struct encoder *_encoder)
}
static bool
-flac_encoder_setup(struct flac_encoder *encoder, GError **error)
+flac_encoder_setup(struct flac_encoder *encoder, unsigned bits_per_sample,
+ GError **error)
{
if ( !FLAC__stream_encoder_set_compression_level(encoder->fse,
encoder->compression)) {
@@ -106,10 +107,10 @@ flac_encoder_setup(struct flac_encoder *encoder, GError **error)
return false;
}
if ( !FLAC__stream_encoder_set_bits_per_sample(encoder->fse,
- encoder->audio_format.bits)) {
+ bits_per_sample)) {
g_set_error(error, flac_encoder_quark(), 0,
"error setting flac bit format to %d",
- encoder->audio_format.bits);
+ bits_per_sample);
return false;
}
if ( !FLAC__stream_encoder_set_sample_rate(encoder->fse,
@@ -143,13 +144,29 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
GError **error)
{
struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
+ unsigned bits_per_sample;
FLAC__StreamEncoderInitStatus init_status;
encoder->audio_format = *audio_format;
/* FIXME: flac should support 32bit as well */
- if (audio_format->bits > 24)
- audio_format->bits = 24;
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ bits_per_sample = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ bits_per_sample = 16;
+ break;
+
+ case SAMPLE_FORMAT_S24_P32:
+ bits_per_sample = 24;
+ break;
+
+ default:
+ bits_per_sample = 24;
+ audio_format->format = SAMPLE_FORMAT_S24_P32;
+ }
/* allocate the encoder */
encoder->fse = FLAC__stream_encoder_new();
@@ -159,7 +176,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
return false;
}
- if (!flac_encoder_setup(encoder, error)) {
+ if (!flac_encoder_setup(encoder, bits_per_sample, error)) {
FLAC__stream_encoder_delete(encoder->fse);
return false;
}
@@ -237,20 +254,23 @@ flac_encoder_write(struct encoder *_encoder,
num_frames = length / audio_format_frame_size(&encoder->audio_format);
num_samples = num_frames * encoder->audio_format.channels;
- switch (encoder->audio_format.bits) {
- case 8:
+ switch (encoder->audio_format.format) {
+ case SAMPLE_FORMAT_S8:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*4);
pcm8_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
- case 16:
+
+ case SAMPLE_FORMAT_S16:
exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*2);
pcm16_to_flac(exbuffer, data, num_samples);
buffer = exbuffer;
break;
- case 24:
- case 32: /* nothing need to be done
- * format is the same for both mpd and libFLAC */
+
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
+ /* nothing need to be done; format is the same for
+ both mpd and libFLAC */
buffer = data;
break;
}
diff --git a/src/encoder/lame_encoder.c b/src/encoder/lame_encoder.c
index 812ff39c5..97431a817 100644
--- a/src/encoder/lame_encoder.c
+++ b/src/encoder/lame_encoder.c
@@ -185,7 +185,7 @@ lame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
{
struct lame_encoder *encoder = (struct lame_encoder *)_encoder;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/twolame_encoder.c b/src/encoder/twolame_encoder.c
index cddf5773e..e7af89bf6 100644
--- a/src/encoder/twolame_encoder.c
+++ b/src/encoder/twolame_encoder.c
@@ -192,7 +192,7 @@ twolame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format
{
struct twolame_encoder *encoder = (struct twolame_encoder *)_encoder;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
audio_format->channels = 2;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/vorbis_encoder.c b/src/encoder/vorbis_encoder.c
index 2fa0fd950..d072bcd3f 100644
--- a/src/encoder/vorbis_encoder.c
+++ b/src/encoder/vorbis_encoder.c
@@ -212,7 +212,7 @@ vorbis_encoder_open(struct encoder *_encoder,
struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
bool ret;
- audio_format->bits = 16;
+ audio_format->format = SAMPLE_FORMAT_S16;
encoder->audio_format = *audio_format;
diff --git a/src/encoder/wave_encoder.c b/src/encoder/wave_encoder.c
index e66cc1917..f34ae0241 100644
--- a/src/encoder/wave_encoder.c
+++ b/src/encoder/wave_encoder.c
@@ -114,16 +114,39 @@ wave_encoder_open(struct encoder *_encoder,
struct wave_encoder *encoder = (struct wave_encoder *)_encoder;
void *buffer;
- encoder->bits = audio_format->bits;
+ assert(audio_format_valid(audio_format));
+
+ switch (audio_format->format) {
+ case SAMPLE_FORMAT_S8:
+ encoder->bits = 8;
+ break;
+
+ case SAMPLE_FORMAT_S16:
+ encoder->bits = 16;
+ break;
+
+ case SAMPLE_FORMAT_S24_P32:
+ encoder->bits = 24;
+ break;
+
+ case SAMPLE_FORMAT_S32:
+ encoder->bits = 32;
+ break;
+
+ default:
+ audio_format->format = SAMPLE_FORMAT_S16;
+ encoder->bits = 16;
+ break;
+ }
buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
/* create PCM wave header in initial buffer */
fill_wave_header((struct wave_header *) buffer,
audio_format->channels,
- audio_format->bits,
+ encoder->bits,
audio_format->sample_rate,
- (audio_format->bits / 8) * audio_format->channels );
+ (encoder->bits / 8) * audio_format->channels );
encoder->buffer_length = sizeof(struct wave_header);
return true;