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author | Simon Hosie <simonhosie77+mpd@gmail.com> | 2012-09-25 21:08:32 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2012-09-25 21:27:50 +0200 |
commit | 5e9ccdec639a7b3e6b3e25c1fcac5270f8c5005c (patch) | |
tree | b5c8df7f5071cbf95e42bc284c64ac8c977bcc75 | |
parent | 071aca60be8ddeff508d9331bd78e6aa87d9fc52 (diff) | |
download | mpd-5e9ccdec639a7b3e6b3e25c1fcac5270f8c5005c.tar.gz mpd-5e9ccdec639a7b3e6b3e25c1fcac5270f8c5005c.tar.xz mpd-5e9ccdec639a7b3e6b3e25c1fcac5270f8c5005c.zip |
decoder/vorbis: skip 16 bit quantisation, provide float samples
Internally the vorbis (non-Tremor) decoder is working in floating
point, and it's not really necessary to cut the output back to 16-bit
if the soundcard or OS supports higher resolution.
The decoder can be trivially modified to bypass its internal
quantisation and produce floating-point output, and a separate
quantisation can be used as appropriate to the platform.
-rw-r--r-- | NEWS | 1 | ||||
-rw-r--r-- | src/decoder/vorbis_decoder_plugin.c | 44 |
2 files changed, 45 insertions, 0 deletions
@@ -1,6 +1,7 @@ ver 0.18 (2012/??/??) * decoder: - opus: new decoder plugin for the Opus codec + - vorbis: skip 16 bit quantisation, provide float samples * improved decoder/output error reporting ver 0.17.2 (2012/??/??) diff --git a/src/decoder/vorbis_decoder_plugin.c b/src/decoder/vorbis_decoder_plugin.c index ff6e76eb7..f180a6559 100644 --- a/src/decoder/vorbis_decoder_plugin.c +++ b/src/decoder/vorbis_decoder_plugin.c @@ -161,6 +161,21 @@ vorbis_send_comments(struct decoder *decoder, struct input_stream *is, tag_free(tag); } +#ifndef HAVE_TREMOR +static void +vorbis_interleave(float *dest, const float *const*src, + unsigned nframes, unsigned channels) +{ + for (const float *const*src_end = src + channels; + src != src_end; ++src, ++dest) { + float *d = dest; + for (const float *s = *src, *s_end = s + nframes; + s != s_end; ++s, d += channels) + *d = *s; + } +} +#endif + /* public */ static void vorbis_stream_decode(struct decoder *decoder, @@ -188,7 +203,11 @@ vorbis_stream_decode(struct decoder *decoder, struct audio_format audio_format; if (!audio_format_init_checked(&audio_format, vi->rate, +#ifdef HAVE_TREMOR SAMPLE_FORMAT_S16, +#else + SAMPLE_FORMAT_FLOAT, +#endif vi->channels, &error)) { g_warning("%s", error->message); g_error_free(error); @@ -202,7 +221,16 @@ vorbis_stream_decode(struct decoder *decoder, decoder_initialized(decoder, &audio_format, vis.seekable, total_time); enum decoder_command cmd = decoder_get_command(decoder); + +#ifdef HAVE_TREMOR char buffer[4096]; +#else + float buffer[2048]; + const int frames_per_buffer = + G_N_ELEMENTS(buffer) / audio_format.channels; + const unsigned frame_size = sizeof(buffer[0]) * audio_format.channels; +#endif + int prev_section = -1; unsigned kbit_rate = 0; @@ -216,9 +244,25 @@ vorbis_stream_decode(struct decoder *decoder, } int current_section; + +#ifdef HAVE_TREMOR long nbytes = ov_read(&vf, buffer, sizeof(buffer), VORBIS_BIG_ENDIAN, 2, 1, ¤t_section); +#else + float **per_channel; + long nframes = ov_read_float(&vf, &per_channel, + frames_per_buffer, + ¤t_section); + long nbytes = nframes; + if (nframes > 0) { + vorbis_interleave(buffer, + (const float*const*)per_channel, + nframes, audio_format.channels); + nbytes *= frame_size; + } +#endif + if (nbytes == OV_HOLE) /* bad packet */ nbytes = 0; else if (nbytes <= 0) |