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author | Max Kellermann <max@duempel.org> | 2012-03-21 19:19:40 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2012-03-21 19:25:59 +0100 |
commit | 1c84f324a13bcc3d7fbd84f8d59398b1c801247a (patch) | |
tree | b927c7273322b20eec2285820796d46181d92f31 /src/output | |
parent | 3dba09f339407716a20054b34e043e47470c645f (diff) | |
download | mpd-1c84f324a13bcc3d7fbd84f8d59398b1c801247a.tar.gz mpd-1c84f324a13bcc3d7fbd84f8d59398b1c801247a.tar.xz mpd-1c84f324a13bcc3d7fbd84f8d59398b1c801247a.zip |
output/oss: always receive host byte order samples
Don't use audio_format.reverse_endian.
Diffstat (limited to 'src/output')
-rw-r--r-- | src/output/oss_output_plugin.c | 75 |
1 files changed, 68 insertions, 7 deletions
diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c index 2309f0f3c..ea15022d9 100644 --- a/src/output/oss_output_plugin.c +++ b/src/output/oss_output_plugin.c @@ -51,9 +51,23 @@ #undef AFMT_S24_NE #endif +#ifdef AFMT_S24_PACKED +#include "pcm_buffer.h" +#include "pcm_byteswap.h" +#endif + struct oss_data { struct audio_output base; +#ifdef AFMT_S24_PACKED + /** + * The buffer used to reverse the byte order. + * + * @see #reverse_endian + */ + struct pcm_buffer reverse_buffer; +#endif + int fd; const char *device; @@ -62,6 +76,16 @@ struct oss_data { * the device after cancel(). */ struct audio_format audio_format; + +#ifdef AFMT_S24_PACKED + /** + * Does OSS expect samples in reverse byte order? (i.e. not + * host byte order) + * + * This attribute is only valid while the device is open. + */ + bool reverse_endian; +#endif }; /** @@ -221,6 +245,27 @@ oss_output_finish(struct audio_output *ao) oss_data_free(od); } +#ifdef AFMT_S24_PACKED + +static bool +oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r) +{ + struct oss_data *od = (struct oss_data *)ao; + + pcm_buffer_init(&od->reverse_buffer); + return true; +} + +static void +oss_output_disable(struct audio_output *ao) +{ + struct oss_data *od = (struct oss_data *)ao; + + pcm_buffer_deinit(&od->reverse_buffer); +} + +#endif + static void oss_close(struct oss_data *od) { @@ -471,6 +516,9 @@ sample_format_from_oss(int format) */ static bool oss_setup_sample_format(int fd, struct audio_format *audio_format, +#ifdef AFMT_S24_PACKED + bool *reverse_endian_r, +#endif GError **error_r) { const char *const msg = "Failed to set sample format"; @@ -489,9 +537,8 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, audio_format->format = mpd_format; #ifdef AFMT_S24_PACKED - if (oss_format == AFMT_S24_PACKED) - audio_format->reverse_endian = - G_BYTE_ORDER != G_LITTLE_ENDIAN; + *reverse_endian_r = oss_format == AFMT_S24_PACKED && + G_BYTE_ORDER != G_LITTLE_ENDIAN; #endif return true; @@ -536,9 +583,8 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format, audio_format->format = mpd_format; #ifdef AFMT_S24_PACKED - if (oss_format == AFMT_S24_PACKED) - audio_format->reverse_endian = - G_BYTE_ORDER != G_LITTLE_ENDIAN; + *reverse_endian_r = oss_format == AFMT_S24_PACKED && + G_BYTE_ORDER != G_LITTLE_ENDIAN; #endif return true; @@ -563,7 +609,11 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format, { return oss_setup_channels(od->fd, audio_format, error_r) && oss_setup_sample_rate(od->fd, audio_format, error_r) && - oss_setup_sample_format(od->fd, audio_format, error_r); + oss_setup_sample_format(od->fd, audio_format, +#ifdef AFMT_S24_PACKED + &od->reverse_endian, +#endif + error_r); } /** @@ -675,6 +725,13 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size, if (od->fd < 0 && !oss_reopen(od, error)) return 0; +#ifdef AFMT_S24_PACKED + if (od->reverse_endian) + chunk = pcm_byteswap(&od->reverse_buffer, + od->audio_format.format, + chunk, size); +#endif + while (true) { ret = write(od->fd, chunk, size); if (ret > 0) @@ -694,6 +751,10 @@ const struct audio_output_plugin oss_output_plugin = { .test_default_device = oss_output_test_default_device, .init = oss_output_init, .finish = oss_output_finish, +#ifdef AFMT_S24_PACKED + .enable = oss_output_enable, + .disable = oss_output_disable, +#endif .open = oss_output_open, .close = oss_output_close, .play = oss_output_play, |