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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/decoder/_flac_common.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 'src/decoder/_flac_common.c')
-rw-r--r-- | src/decoder/_flac_common.c | 27 |
1 files changed, 25 insertions, 2 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c index f12b8bff0..70b2c0202 100644 --- a/src/decoder/_flac_common.c +++ b/src/decoder/_flac_common.c @@ -60,6 +60,27 @@ flac_data_deinit(struct flac_data *data) tag_free(data->tag); } +static enum sample_format +flac_sample_format(const FLAC__StreamMetadata_StreamInfo *si) +{ + switch (si->bits_per_sample) { + case 8: + return SAMPLE_FORMAT_S8; + + case 16: + return SAMPLE_FORMAT_S16; + + case 24: + return SAMPLE_FORMAT_S24_P32; + + case 32: + return SAMPLE_FORMAT_S32; + + default: + return SAMPLE_FORMAT_UNDEFINED; + } +} + bool flac_data_get_audio_format(struct flac_data *data, struct audio_format *audio_format) @@ -71,9 +92,11 @@ flac_data_get_audio_format(struct flac_data *data, return false; } + data->sample_format = flac_sample_format(&data->stream_info); + if (!audio_format_init_checked(audio_format, data->stream_info.sample_rate, - data->stream_info.bits_per_sample, + data->sample_format, data->stream_info.channels, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -144,7 +167,7 @@ flac_common_write(struct flac_data *data, const FLAC__Frame * frame, buffer = pcm_buffer_get(&data->buffer, buffer_size); flac_convert(buffer, frame->header.channels, - frame->header.bits_per_sample, buf, + data->sample_format, buf, 0, frame->header.blocksize); if (data->next_frame >= data->first_frame) |