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osxaudio: use GST_AUDIO_INFO_* accessors

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Changes include the following:

 * Update classname references
 * Replace GST_BOILERPLATE_FULL with G_DEFINE_TYPE
 * Use new GstAudioInfo struct and methods
 * Use new buffer memory allocation scheme

Conflicts:
	sys/osxaudio/gstosxaudioringbuffer.c
This commit is contained in:
Todd Agulnick 2013-04-12 12:18:04 -07:00 committed by Sebastian Dröge
parent ed94ef79f9
commit c2c85a094a
1 changed files with 13 additions and 13 deletions
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26
sys/osxaudio/gstosxaudioringbuffer.c
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@ -178,7 +178,7 @@ gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
if (RINGBUFFER_IS_SPDIF (spec->type)) { if (RINGBUFFER_IS_SPDIF (spec->type)) {
format.mFormatID = kAudioFormat60958AC3; format.mFormatID = kAudioFormat60958AC3;
format.mSampleRate = (double) spec->info.rate; format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = 2; format.mChannelsPerFrame = 2;
format.mFormatFlags = kAudioFormatFlagIsSignedInteger | format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable; kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
@ -194,32 +194,32 @@ gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
int width, depth; int width, depth;
/* Fill out the audio description we're going to be using */ /* Fill out the audio description we're going to be using */
format.mFormatID = kAudioFormatLinearPCM; format.mFormatID = kAudioFormatLinearPCM;
format.mSampleRate = (double) spec->info.rate; format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = spec->info.channels; format.mChannelsPerFrame = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (spec->type == GST_AUDIO_FORMAT_F32) { if (GST_AUDIO_INFO_IS_FLOAT (&spec->info)) {
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked; format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
width = depth = spec->info.finfo->width; width = depth = GST_AUDIO_INFO_WIDTH (&spec->info);
} else { } else {
format.mFormatFlags = kAudioFormatFlagIsSignedInteger; format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
width = spec->info.finfo->width; width = GST_AUDIO_INFO_WIDTH (&spec->info);
depth = spec->info.finfo->depth; depth = GST_AUDIO_INFO_DEPTH (&spec->info);
if (width == depth) { if (width == depth) {
format.mFormatFlags |= kAudioFormatFlagIsPacked; format.mFormatFlags |= kAudioFormatFlagIsPacked;
} else { } else {
format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh; format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
} }
if (spec->info.finfo->endianness == G_BIG_ENDIAN) { if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&spec->info)) {
format.mFormatFlags |= kAudioFormatFlagIsBigEndian; format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
} }
} }
format.mBytesPerFrame = spec->info.channels * (width >> 3); format.mBytesPerFrame = GST_AUDIO_INFO_BPF (&spec->info);
format.mBitsPerChannel = depth; format.mBitsPerChannel = depth;
format.mBytesPerPacket = spec->info.channels * (width >> 3); format.mBytesPerPacket = GST_AUDIO_INFO_BPF (&spec->info);
format.mFramesPerPacket = 1; format.mFramesPerPacket = 1;
format.mReserved = 0; format.mReserved = 0;
spec->segsize = spec->segsize =
(spec->latency_time * spec->info.rate / G_USEC_PER_SEC) * (spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
spec->info.bpf; G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time; spec->segtotal = spec->buffer_time / spec->latency_time;
is_passthrough = FALSE; is_passthrough = FALSE;
} }
@ -301,7 +301,7 @@ gst_osx_audio_ring_buffer_delay (GstAudioRingBuffer * buf)
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf); osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (!gst_core_audio_get_samples_and_latency (osxbuf->core_audio, if (!gst_core_audio_get_samples_and_latency (osxbuf->core_audio,
GST_AUDIO_RING_BUFFER (buf)->spec.info.rate, &samples, &latency)) { GST_AUDIO_INFO_RATE (&buf->spec.info), &samples, &latency)) {
return 0; return 0;
} }
GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples", GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples",