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| author | Max Kellermann <max@duempel.org> | 2008-08-26 08:27:05 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2008-08-26 08:27:05 +0200 |
| commit | 4590a98f0eb9484e185e7e0c25a3373c8e9076ea (patch) | |
| tree | 94ea0d53bdf0b7f9b81ff679b0fb4466d971b1e5 /src/inputPlugins/audiofile_plugin.c | |
| parent | 0d45870cea6836cd48e6953f4e67756b2502e22c (diff) | |
| download | mpd-4590a98f0eb9484e185e7e0c25a3373c8e9076ea.tar.gz mpd-4590a98f0eb9484e185e7e0c25a3373c8e9076ea.tar.xz mpd-4590a98f0eb9484e185e7e0c25a3373c8e9076ea.zip | |
added audio_format parameter to decoder_initialized()
dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().
Diffstat (limited to 'src/inputPlugins/audiofile_plugin.c')
| -rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 21 |
1 files changed, 11 insertions, 10 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c index 696621aff..f96ea2fca 100644 --- a/src/inputPlugins/audiofile_plugin.c +++ b/src/inputPlugins/audiofile_plugin.c @@ -45,6 +45,7 @@ static int audiofile_decode(struct decoder * decoder, char *path) int fs, frame_count; AFfilehandle af_fp; int bits; + AudioFormat audio_format; mpd_uint16 bitRate; struct stat st; @@ -62,30 +63,30 @@ static int audiofile_decode(struct decoder * decoder, char *path) afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - dc.audioFormat.bits = (mpd_uint8)bits; - dc.audioFormat.sampleRate = + audio_format.bits = (mpd_uint8)bits; + audio_format.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); - dc.audioFormat.channels = + audio_format.channels = (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); - getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat)); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); dc.totalTime = - ((float)frame_count / (float)dc.audioFormat.sampleRate); + ((float)frame_count / (float)audio_format.sampleRate); bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5); - if (dc.audioFormat.bits != 8 && dc.audioFormat.bits != 16) { + if (audio_format.bits != 8 && audio_format.bits != 16) { ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", - path, dc.audioFormat.bits); + path, audio_format.bits); afCloseFile(af_fp); return -1; } fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); - decoder_initialized(decoder); + decoder_initialized(decoder, &audio_format); + { int ret, eof = 0, current = 0; char chunk[CHUNK_SIZE]; @@ -94,7 +95,7 @@ static int audiofile_decode(struct decoder * decoder, char *path) if (dc.command == DECODE_COMMAND_SEEK) { decoder_clear(decoder); current = dc.seekWhere * - dc.audioFormat.sampleRate; + audio_format.sampleRate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); dc_command_finished(); } @@ -110,7 +111,7 @@ static int audiofile_decode(struct decoder * decoder, char *path) 1, chunk, ret * fs, (float)current / - (float)dc.audioFormat. + (float)audio_format. sampleRate, bitRate, NULL); if (dc.command == DECODE_COMMAND_STOP) |