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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/output/alsa_plugin.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/output/alsa_plugin.c')
-rw-r--r-- | src/output/alsa_plugin.c | 73 |
1 files changed, 44 insertions, 29 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 2c642015d..b7325de07 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -185,13 +185,22 @@ alsa_test_default_device(void) static snd_pcm_format_t get_bitformat(const struct audio_format *af) { - switch (af->bits) { - case 8: return SND_PCM_FORMAT_S8; - case 16: return SND_PCM_FORMAT_S16; - case 24: return SND_PCM_FORMAT_S24; - case 32: return SND_PCM_FORMAT_S32; + switch (af->format) { + case SAMPLE_FORMAT_S8: + return SND_PCM_FORMAT_S8; + + case SAMPLE_FORMAT_S16: + return SND_PCM_FORMAT_S16; + + case SAMPLE_FORMAT_S24_P32: + return SND_PCM_FORMAT_S24; + + case SAMPLE_FORMAT_S32: + return SND_PCM_FORMAT_S32; + + default: + return SND_PCM_FORMAT_UNKNOWN; } - return SND_PCM_FORMAT_UNKNOWN; } static snd_pcm_format_t @@ -264,61 +273,67 @@ configure_hw: err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, byteswap_bitformat(bitformat)); if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n", - alsa_device(ad), audio_format->bits); + g_debug("ALSA device \"%s\": converting format %s to reverse-endian", + alsa_device(ad), + sample_format_to_string(audio_format->format)); audio_format->reverse_endian = 1; } } - if (err == -EINVAL && (audio_format->bits == 24 || - audio_format->bits == 16)) { + if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 || + audio_format->format == SAMPLE_FORMAT_S16)) { /* fall back to 32 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, SND_PCM_FORMAT_S32); if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n", - alsa_device(ad), audio_format->bits); - audio_format->bits = 32; + g_debug("ALSA device \"%s\": converting format %s to 32 bit\n", + alsa_device(ad), + sample_format_to_string(audio_format->format)); + audio_format->format = SAMPLE_FORMAT_S32; } } - if (err == -EINVAL && (audio_format->bits == 24 || - audio_format->bits == 16)) { + if (err == -EINVAL && (audio_format->format == SAMPLE_FORMAT_S24_P32 || + audio_format->format == SAMPLE_FORMAT_S16)) { /* fall back to 32 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, byteswap_bitformat(SND_PCM_FORMAT_S32)); if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n", - alsa_device(ad), audio_format->bits); - audio_format->bits = 32; + g_debug("ALSA device \"%s\": converting format %s to 32 bit backward-endian\n", + alsa_device(ad), + sample_format_to_string(audio_format->format)); + audio_format->format = SAMPLE_FORMAT_S32; audio_format->reverse_endian = 1; } } - if (err == -EINVAL && audio_format->bits != 16) { + if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) { /* fall back to 16 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, SND_PCM_FORMAT_S16); if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n", - alsa_device(ad), audio_format->bits); - audio_format->bits = 16; + g_debug("ALSA device \"%s\": converting format %s to 16 bit\n", + alsa_device(ad), + sample_format_to_string(audio_format->format)); + audio_format->format = SAMPLE_FORMAT_S16; } } - if (err == -EINVAL && audio_format->bits != 16) { + if (err == -EINVAL && audio_format->format != SAMPLE_FORMAT_S16) { /* fall back to 16 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, byteswap_bitformat(SND_PCM_FORMAT_S16)); if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n", - alsa_device(ad), audio_format->bits); - audio_format->bits = 16; + g_debug("ALSA device \"%s\": converting format %s to 16 bit backward-endian\n", + alsa_device(ad), + sample_format_to_string(audio_format->format)); + audio_format->format = SAMPLE_FORMAT_S16; audio_format->reverse_endian = 1; } } if (err < 0) { g_set_error(error, alsa_output_quark(), err, - "ALSA device \"%s\" does not support %u bit audio: %s", - alsa_device(ad), audio_format->bits, + "ALSA device \"%s\" does not support format %s: %s", + alsa_device(ad), + sample_format_to_string(audio_format->format), snd_strerror(-err)); return false; } @@ -449,7 +464,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error) /* sample format is not supported by this plugin - fall back to 16 bit samples */ - audio_format->bits = 16; + audio_format->format = SAMPLE_FORMAT_S16; bitformat = SND_PCM_FORMAT_S16; } |