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author | Max Kellermann <max@duempel.org> | 2008-10-10 14:41:37 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2008-10-10 14:41:37 +0200 |
commit | 96155a3376ef40eac9ca7a7b882da6447ab53979 (patch) | |
tree | b30e5248b366b12ef860c65ea3640598f3b73658 /src/audioOutputs | |
parent | de2cb3f37568e7680549057f8d7b6d748c388480 (diff) | |
download | mpd-96155a3376ef40eac9ca7a7b882da6447ab53979.tar.gz mpd-96155a3376ef40eac9ca7a7b882da6447ab53979.tar.xz mpd-96155a3376ef40eac9ca7a7b882da6447ab53979.zip |
audio_format: added audio_format_frame_size()
A frame contains one sample per channel, thus it is sample_size *
channels. This patch includes some cleanup for various locations
where the sample size for 24 bit audio was still 3 bytes (instead of
4).
Diffstat (limited to 'src/audioOutputs')
-rw-r--r-- | src/audioOutputs/audioOutput_alsa.c | 2 | ||||
-rw-r--r-- | src/audioOutputs/audioOutput_osx.c | 5 |
2 files changed, 3 insertions, 4 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c index 30ad449f3..9d47ff244 100644 --- a/src/audioOutputs/audioOutput_alsa.c +++ b/src/audioOutputs/audioOutput_alsa.c @@ -288,7 +288,7 @@ configure_hw: if (err < 0) goto error; - ad->sampleSize = audio_format_sample_size(audioFormat) * audioFormat->channels; + ad->sampleSize = audio_format_frame_size(audioFormat); DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at " "%u Hz\n", ad->device, audioFormat->bits, diff --git a/src/audioOutputs/audioOutput_osx.c b/src/audioOutputs/audioOutput_osx.c index 1fc0a5d9e..65060cc8c 100644 --- a/src/audioOutputs/audioOutput_osx.c +++ b/src/audioOutputs/audioOutput_osx.c @@ -266,8 +266,7 @@ static int osx_openDevice(struct audio_output *audioOutput, streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; #endif - streamDesc.mBytesPerPacket = - audioFormat->channels * audio_format_sample_size(audioFormat); + streamDesc.mBytesPerPacket = audio_format_frame_size(audioFormat); streamDesc.mFramesPerPacket = 1; streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket; streamDesc.mChannelsPerFrame = audioFormat->channels; @@ -284,7 +283,7 @@ static int osx_openDevice(struct audio_output *audioOutput, /* create a buffer of 1s */ od->bufferSize = (audioFormat->sample_rate) * - (audioFormat->bits >> 3) * (audioFormat->channels); + audio_format_frame_size(audioFormat); od->buffer = xrealloc(od->buffer, od->bufferSize); od->pos = 0; |