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author | Max Kellermann <max@duempel.org> | 2011-10-08 15:36:29 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2011-10-20 02:55:05 +0200 |
commit | 8465c5fe0e154b96edea992fdcb574060934b9e5 (patch) | |
tree | 243362cb3db60395f4bdb29170f096b73b3a8937 /src/pcm_convert.c | |
parent | 42e248a8da0f2043e752166d4a34615c850384ec (diff) | |
download | mpd-8465c5fe0e154b96edea992fdcb574060934b9e5.tar.gz mpd-8465c5fe0e154b96edea992fdcb574060934b9e5.tar.xz mpd-8465c5fe0e154b96edea992fdcb574060934b9e5.zip |
pcm_format: implement conversion to float
Diffstat (limited to '')
-rw-r--r-- | src/pcm_convert.c | 69 |
1 files changed, 68 insertions, 1 deletions
diff --git a/src/pcm_convert.c b/src/pcm_convert.c index 58137fd0e..daf1cd234 100644 --- a/src/pcm_convert.c +++ b/src/pcm_convert.c @@ -24,6 +24,7 @@ #include "pcm_byteswap.h" #include "pcm_pack.h" #include "audio_format.h" +#include "glib_compat.h" #include <assert.h> #include <string.h> @@ -56,7 +57,6 @@ void pcm_convert_deinit(struct pcm_convert_state *state) pcm_buffer_deinit(&state->byteswap_buffer); } -G_GNUC_UNUSED static const void * pcm_convert_channels(struct pcm_buffer *buffer, enum sample_format format, uint8_t dest_channels, @@ -312,6 +312,67 @@ pcm_convert_32(struct pcm_convert_state *state, return buf; } +static const float * +pcm_convert_float(struct pcm_convert_state *state, + const struct audio_format *src_format, + const void *src_buffer, size_t src_size, + const struct audio_format *dest_format, size_t *dest_size_r, + GError **error_r) +{ + const float *buffer = src_buffer; + size_t size = src_size; + + assert(dest_format->format == SAMPLE_FORMAT_FLOAT); + + if (src_format->reverse_endian || dest_format->reverse_endian) { + g_set_error_literal(error_r, pcm_convert_quark(), 0, + "Reverse endian not supported"); + return NULL; + } + + /* convert channels first, hoping the source format is + supported (float is not) */ + + if (dest_format->channels != src_format->channels) { + buffer = pcm_convert_channels(&state->channels_buffer, + src_format->format, + dest_format->channels, + src_format->channels, + buffer, size, &size, error_r); + if (buffer == NULL) + return NULL; + } + + /* convert to float now */ + + buffer = pcm_convert_to_float(&state->format_buffer, + src_format->format, + buffer, size, &size); + if (buffer == NULL) { + g_set_error(error_r, pcm_convert_quark(), 0, + "Conversion from %s to float is not implemented", + sample_format_to_string(src_format->format)); + return NULL; + } + + /* resample with float, because this is the best format for + libsamplerate */ + + if (src_format->sample_rate != dest_format->sample_rate) { + buffer = pcm_resample_float(&state->resample, + dest_format->channels, + src_format->sample_rate, + buffer, size, + dest_format->sample_rate, &size, + error_r); + if (buffer == NULL) + return NULL; + } + + *dest_size_r = size; + return buffer; +} + const void * pcm_convert(struct pcm_convert_state *state, const struct audio_format *src_format, @@ -359,6 +420,12 @@ pcm_convert(struct pcm_convert_state *state, dest_format, dest_size_r, error_r); + case SAMPLE_FORMAT_FLOAT: + return pcm_convert_float(state, + src_format, src, src_size, + dest_format, dest_size_r, + error_r); + default: g_set_error(error_r, pcm_convert_quark(), 0, "PCM conversion to %s is not implemented", |