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author | Eric Wong <normalperson@yhbt.net> | 2008-04-13 01:16:03 +0000 |
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committer | Eric Wong <normalperson@yhbt.net> | 2008-04-13 01:16:03 +0000 |
commit | dec6b1612e953c6029d963ff55d2b4a669b60f43 (patch) | |
tree | a1138cb07f67c821ee5000618302d21367ab2245 /src/inputPlugins/aac_plugin.c | |
parent | 98acfa8ac5bac09ca49a7c21938b5a5801e01ca5 (diff) | |
download | mpd-dec6b1612e953c6029d963ff55d2b4a669b60f43.tar.gz mpd-dec6b1612e953c6029d963ff55d2b4a669b60f43.tar.xz mpd-dec6b1612e953c6029d963ff55d2b4a669b60f43.zip |
Stop passing our single DecoderControl object everywhere
This at least makes the argument list to a lot of our plugin
functions shorter and removes a good amount of line nois^W^Wcode,
hopefully making things easier to read and follow.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7353 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r-- | src/inputPlugins/aac_plugin.c | 34 |
1 files changed, 17 insertions, 17 deletions
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c index 2962b57c6..aeda10492 100644 --- a/src/inputPlugins/aac_plugin.c +++ b/src/inputPlugins/aac_plugin.c @@ -282,7 +282,7 @@ static int getAacTotalTime(char *file) return file_time; } -static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path) +static int aac_decode(OutputBuffer * cb, char *path) { float file_time; float totalTime; @@ -339,9 +339,9 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path) return -1; } - dc->audioFormat.bits = 16; + dc.audioFormat.bits = 16; - dc->totalTime = totalTime; + dc.totalTime = totalTime; file_time = 0.0; @@ -372,12 +372,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path) sampleRate = frameInfo.samplerate; #endif - if (dc->state != DECODE_STATE_DECODE) { - dc->audioFormat.channels = frameInfo.channels; - dc->audioFormat.sampleRate = sampleRate; - getOutputAudioFormat(&(dc->audioFormat), + if (dc.state != DECODE_STATE_DECODE) { + dc.audioFormat.channels = frameInfo.channels; + dc.audioFormat.sampleRate = sampleRate; + getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat)); - dc->state = DECODE_STATE_DECODE; + dc.state = DECODE_STATE_DECODE; } advanceAacBuffer(&b, frameInfo.bytesconsumed); @@ -395,14 +395,14 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path) sampleBufferLen = sampleCount * 2; - sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer, + sendDataToOutputBuffer(cb, NULL, 0, sampleBuffer, sampleBufferLen, file_time, bitRate, NULL); - if (dc->seek) { - dc->seekError = 1; - dc->seek = 0; + if (dc.seek) { + dc.seekError = 1; + dc.seek = 0; decoder_wakeup_player(); - } else if (dc->stop) { + } else if (dc.stop) { eof = 1; break; } @@ -414,12 +414,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path) if (b.buffer) free(b.buffer); - if (dc->state != DECODE_STATE_DECODE) + if (dc.state != DECODE_STATE_DECODE) return -1; - if (dc->seek) { - dc->seekError = 1; - dc->seek = 0; + if (dc.seek) { + dc.seekError = 1; + dc.seek = 0; decoder_wakeup_player(); } |