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author | Max Kellermann <max@duempel.org> | 2009-11-10 17:11:34 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2009-12-02 22:29:50 +0100 |
commit | c412d6251e9cd3abe735b7622af4003502e54f72 (patch) | |
tree | 7344c13f62e4cc788c830c05d21bb7b5b47f5866 /src/encoder/flac_encoder.c | |
parent | 68c2cfbb4067b2292e1ff1d4e7716ff370903f84 (diff) | |
download | mpd-c412d6251e9cd3abe735b7622af4003502e54f72.tar.gz mpd-c412d6251e9cd3abe735b7622af4003502e54f72.tar.xz mpd-c412d6251e9cd3abe735b7622af4003502e54f72.zip |
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 '')
-rw-r--r-- | src/encoder/flac_encoder.c | 44 |
1 files changed, 32 insertions, 12 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; } |