Mercurial > audlegacy-plugins
changeset 2360:ead24454f4b7
native 24bit output for FLAC
| author | Eugene Zagidullin <e.asphyx@gmail.com> |
|---|---|
| date | Tue, 05 Feb 2008 07:37:00 +0300 |
| parents | b3475063c000 |
| children | c831816c0b87 2b561aa49580 |
| files | src/flacng/flacng.h src/flacng/plugin.c src/flacng/seekable_stream_callbacks.c |
| diffstat | 3 files changed, 33 insertions(+), 76 deletions(-) [+] |
line wrap: on
line diff
--- a/src/flacng/flacng.h Mon Feb 04 23:12:46 2008 +0300 +++ b/src/flacng/flacng.h Tue Feb 05 07:37:00 2008 +0300 @@ -30,6 +30,9 @@ #define BUFFER_SIZE_SAMP (FLAC__MAX_BLOCK_SIZE * FLAC__MAX_CHANNELS) #define BUFFER_SIZE_BYTE (BUFFER_SIZE_SAMP * (FLAC__MAX_BITS_PER_SAMPLE/8)) +#define SAMPLE_SIZE(a) (a == 8 ? sizeof(guint8) : (a == 16 ? sizeof(guint16) : sizeof(guint32))) +#define SAMPLE_FMT(a) (a == 8 ? FMT_S8 : (a == 16 ? FMT_S16_NE : (a == 24 ? FMT_S24_NE : FMT_S32_NE))) + /* * Audio information about the stream as a whole, gathered from * the metadata header
--- a/src/flacng/plugin.c Mon Feb 04 23:12:46 2008 +0300 +++ b/src/flacng/plugin.c Tue Feb 05 07:37:00 2008 +0300 @@ -16,6 +16,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ +/* #define FLACNG_DEBUG */ #include "flacng.h" #include <audacious/util.h> @@ -216,9 +217,10 @@ _LEAVE FALSE; } - if ((16 != test_info->stream.bits_per_sample) && + if ((8 != test_info->stream.bits_per_sample) && + (16 != test_info->stream.bits_per_sample) && (24 != test_info->stream.bits_per_sample) && - (8 != test_info->stream.bits_per_sample)) { + (32 != test_info->stream.bits_per_sample)) { _ERROR("This number of bits (%d) is currently not supported, stream not handled by this plugin", test_info->stream.bits_per_sample); INFO_UNLOCK(test_info); @@ -264,12 +266,7 @@ /* --- */ -void squeeze_audio(gint32* src, void* dst, guint count, guint src_res, guint dst_res) { - - /* - * Changes the sample width of count samples in src from - * src_res to dst_res and places the result in dst - */ +void squeeze_audio(gint32* src, void* dst, guint count, guint res) { gint i; gint32* rp = src; @@ -279,68 +276,25 @@ _ENTER; - _DEBUG("Converting %d samples %d->%d", count, src_res, dst_res); + _DEBUG("Converting %d samples to %d bit", count, res); - if ((src_res % 8 != 0) || (dst_res % 8 != 0)) { - _ERROR("Can not convert from %d bps to %d bps: not a multiple of 8", - src_res, dst_res); + if (res % 8 != 0) { + _ERROR("Can not convert to %d bps: not a multiple of 8", res); _LEAVE; } - if (16 == dst_res) { - if (8 == src_res) { - for (i=0; i<count; i++, wp2++, rp++) { - *wp2 = (*rp << 8) & 0xffff; - } - } else if (16 == src_res) { - for (i=0; i<count; i++, wp2++, rp++) { - *wp2 = *rp & 0xffff; - } - } else if (24 == src_res) { - for (i=0; i<count; i++, wp2++, rp++) { - *wp2 = (*rp >> 8) & 0xffff; - } - } else if (32 == src_res) { - for (i=0; i<count; i++, wp2++, rp++) { - *wp2 = (*rp >> 16) & 0xffff; - } + if (res == 8) { + for (i=0; i<count; i++, wp++, rp++) { + *wp = *rp & 0xff; } - } else if (8 == dst_res) { - if (8 == src_res) { - for (i=0; i<count; i++, wp++, rp++) { - *wp = *rp & 0xff; - } - } else if (16 == src_res) { - for (i=0; i<count; i++, wp++, rp++) { - *wp = (*rp >> 8) & 0xff; - } - } else if (24 == src_res) { - for (i=0; i<count; i++, wp++, rp++) { - *wp = (*rp >> 16) & 0xff; - } - } else if (32 == src_res) { - for (i=0; i<count; i++, wp++, rp++) { - *wp = (*rp >> 24) & 0xff; - } + } else if (res == 16) { + for (i=0; i<count; i++, wp2++, rp++) { + *wp2 = *rp & 0xffff; } - } else if (32 == dst_res) { - if (8 == src_res) { - for (i=0; i<count; i++, wp4++, rp++) { - *wp4 = (*rp << 24) & 0xffffffff; - } - } else if (16 == src_res) { - for (i=0; i<count; i++, wp4++, rp++) { - *wp4 = (*rp << 16) & 0xffffffff; - } - } else if (24 == src_res) { - for (i=0; i<count; i++, wp4++, rp++) { - *wp4 = (*rp << 8) & 0xffffffff; - } - } else if (32 == src_res) { - for (i=0; i<count; i++, wp4++, rp++) { - *wp4 = *rp; - } - } + } else if (res == 24 || res == 32) { /* 24bit value stored in lowest 3 bytes */ + for (i=0; i<count; i++, wp4++, rp++) { + *wp4 = *rp; + } } _LEAVE; @@ -377,9 +331,9 @@ stream_info.channels = main_info->stream.channels; main_info->metadata_changed = FALSE; - if (!playback->output->open_audio(FMT_S16_NE, - main_info->stream.samplerate, - main_info->stream.channels)) { + if (!playback->output->open_audio(SAMPLE_FMT(main_info->stream.bits_per_sample), + main_info->stream.samplerate, + main_info->stream.channels)) { playback->playing = FALSE; _ERROR("Could not open output plugin!"); _LEAVE NULL; @@ -454,17 +408,16 @@ sample_count = MIN(OUTPUT_BLOCK_SIZE, elements_left); - /* - * Convert the audio. - * Currently this is hardcoded to 16 bit. - * 8 and 24 bit are sampled up/down linear. - */ - _DEBUG("Converting %d samples to FMT_S16_NE", sample_count); - squeeze_audio(read_pointer, play_buffer, sample_count, main_info->frame.bits_per_sample, 16); + squeeze_audio(read_pointer, play_buffer, sample_count, main_info->stream.bits_per_sample); _DEBUG("Copying %d samples to output plugin", sample_count); - playback->pass_audio(playback, FMT_S16_NE, main_info->frame.channels, sample_count * sizeof(gint16), play_buffer, NULL); + playback->pass_audio(playback, + SAMPLE_FMT(main_info->stream.bits_per_sample), + main_info->stream.channels, + sample_count * SAMPLE_SIZE(main_info->stream.bits_per_sample), + play_buffer, + NULL); read_pointer += sample_count; elements_left -= sample_count;
--- a/src/flacng/seekable_stream_callbacks.c Mon Feb 04 23:12:46 2008 +0300 +++ b/src/flacng/seekable_stream_callbacks.c Tue Feb 05 07:37:00 2008 +0300 @@ -209,7 +209,8 @@ if ((frame->header.bits_per_sample != 8) && (frame->header.bits_per_sample != 16) && - (frame->header.bits_per_sample != 24)) { + (frame->header.bits_per_sample != 24) && + (frame->header.bits_per_sample != 32)) { _ERROR("Unsupported bitrate found in stream: %d!", frame->header.bits_per_sample); _LEAVE FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; }