[vlc-commits] auhal: remove unneeded global variable
David Fuhrmann
git at videolan.org
Mon Feb 24 15:28:21 CET 2014
vlc | branch: master | David Fuhrmann <david.fuhrmann at googlemail.com> | Sun Feb 23 19:32:08 2014 +0100| [de2d863309bb0a1cc797d6d5d43e38562c7c14fe] | committer: David Fuhrmann
auhal: remove unneeded global variable
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=de2d863309bb0a1cc797d6d5d43e38562c7c14fe
---
modules/audio_output/auhal.c | 14 +++++++-------
1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/modules/audio_output/auhal.c b/modules/audio_output/auhal.c
index b6958a4..182d9f7 100644
--- a/modules/audio_output/auhal.c
+++ b/modules/audio_output/auhal.c
@@ -98,7 +98,6 @@ struct aout_sys_t
pid_t i_hog_pid; /* The keep the pid of our hog status */
AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
- AudioStreamBasicDescription stream_format; /* The format we changed the stream to */
AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */
bool b_revert; /* Whether we need to revert the stream format */
bool b_changed_mixing; /* Whether we need to set the mixing mode back */
@@ -716,6 +715,7 @@ static int StartSPDIF(audio_output_t * p_aout, audio_sample_format_t *fmt)
Boolean b_writeable = false;
AudioStreamID *p_streams = NULL;
unsigned i_streams = 0;
+ AudioStreamBasicDescription desired_stream_format;
/* Start doing the SPDIF setup proces */
p_sys->b_digital = true;
@@ -856,11 +856,11 @@ static int StartSPDIF(audio_output_t * p_aout, audio_sample_format_t *fmt)
}
if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio */
- p_sys->stream_format = p_format_list[i_requested_rate_format].mFormat;
+ desired_stream_format = p_format_list[i_requested_rate_format].mFormat;
else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device */
- p_sys->stream_format = p_format_list[i_current_rate_format].mFormat;
+ desired_stream_format = p_format_list[i_current_rate_format].mFormat;
else
- p_sys->stream_format = p_format_list[i_backup_rate_format].mFormat; /* And if we have to, any digital format will be just fine (highest rate possible) */
+ desired_stream_format = p_format_list[i_backup_rate_format].mFormat; /* And if we have to, any digital format will be just fine (highest rate possible) */
}
free(p_format_list);
}
@@ -868,19 +868,19 @@ static int StartSPDIF(audio_output_t * p_aout, audio_sample_format_t *fmt)
msg_Dbg(p_aout, STREAM_FORMAT_MSG("original stream format: ", p_sys->sfmt_revert));
- if (!AudioStreamChangeFormat(p_aout, p_sys->i_stream_id, p_sys->stream_format)) {
+ if (!AudioStreamChangeFormat(p_aout, p_sys->i_stream_id, desired_stream_format)) {
msg_Err(p_aout, "failed to change stream format for SPDIF output");
return false;
}
/* Set the format flags */
- if (p_sys->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
+ if (desired_stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
fmt->i_format = VLC_CODEC_SPDIFB;
else
fmt->i_format = VLC_CODEC_SPDIFL;
fmt->i_bytes_per_frame = AOUT_SPDIF_SIZE;
fmt->i_frame_length = A52_FRAME_NB;
- fmt->i_rate = (unsigned int)p_sys->stream_format.mSampleRate;
+ fmt->i_rate = (unsigned int)desired_stream_format.mSampleRate;
p_sys->i_rate = fmt->i_rate;
aout_FormatPrepare(fmt);
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