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| author | Max Kellermann <max@duempel.org> | 2012-03-27 00:32:10 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2012-03-27 01:17:11 +0200 |
| commit | f6d6110aaaf6ba3e252323c2bf6360ea9782dff2 (patch) | |
| tree | a2a6656d4549620b341f04468fe6fa80ec46ea15 /src | |
| parent | a0730cf264f4bfebffb685f3f08dd8e1072aee38 (diff) | |
| download | mpd-f6d6110aaaf6ba3e252323c2bf6360ea9782dff2.tar.gz mpd-f6d6110aaaf6ba3e252323c2bf6360ea9782dff2.tar.xz mpd-f6d6110aaaf6ba3e252323c2bf6360ea9782dff2.zip | |
output/alsa: move pcm_export_open() to caller
Give the caller more control, prepare for DSD-over-USB improvements.
Diffstat (limited to '')
| -rw-r--r-- | src/output/alsa_output_plugin.c | 27 |
1 files changed, 16 insertions, 11 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c index aaf489ea8..21c3b1d22 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/alsa_output_plugin.c @@ -385,7 +385,7 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, */ static bool alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, - GError **error) + bool *packed_r, bool *reverse_endian_r, GError **error) { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; @@ -429,9 +429,8 @@ configure_hw: ad->writei = snd_pcm_writei; } - bool packed, reverse_endian; err = alsa_output_setup_format(ad->pcm, hwparams, audio_format, - &packed, &reverse_endian); + packed_r, reverse_endian_r); if (err < 0) { g_set_error(error, alsa_output_quark(), err, "ALSA device \"%s\" does not support format %s: %s", @@ -578,9 +577,6 @@ configure_hw: ad->period_frames = alsa_period_size; ad->period_position = 0; - pcm_export_open(&ad->export, audio_format->format, - packed, reverse_endian); - return true; error: @@ -592,7 +588,7 @@ error: static bool alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, - GError **error_r) + bool *packed_r, bool *reverse_endian_r, GError **error_r) { assert(ad->dsd_usb); assert(audio_format->format == SAMPLE_FORMAT_DSD); @@ -604,7 +600,7 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, const struct audio_format check = *audio_format; - if (!alsa_setup(ad, audio_format, error_r)) + if (!alsa_setup(ad, audio_format, packed_r, reverse_endian_r, error_r)) return false; if (!audio_format_equals(audio_format, &check)) { @@ -627,10 +623,19 @@ static bool alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format, GError **error_r) { - if (ad->dsd_usb && audio_format->format == SAMPLE_FORMAT_DSD) - return alsa_setup_dsd(ad, audio_format, error_r); + bool packed, reverse_endian; - return alsa_setup(ad, audio_format, error_r); + const bool success = ad->dsd_usb && audio_format->format == SAMPLE_FORMAT_DSD + ? alsa_setup_dsd(ad, audio_format, &packed, &reverse_endian, + error_r) + : alsa_setup(ad, audio_format, &packed, &reverse_endian, + error_r); + if (!success) + return false; + + pcm_export_open(&ad->export, audio_format->format, + packed, reverse_endian); + return true; } static bool |