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Add format members to AudioConverter for DCHECKing.

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And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2014-11-03 21:32:14 +00:00
parent e451b756a8
commit 5804936052
2 changed files with 15 additions and 5 deletions
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16
webrtc/common_audio/audio_converter.cc
View File

@ -43,10 +43,13 @@ void UpmixFromMono(const float* src,
} // namespace
AudioConverter::AudioConverter(int src_channels, int src_frames,
int dst_channels, int dst_frames) {
int dst_channels, int dst_frames)
: src_channels_(src_channels),
src_frames_(src_frames),
dst_channels_(dst_channels),
dst_frames_(dst_frames) {
CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
const int resample_channels = src_channels < dst_channels ? src_channels :
dst_channels;
const int resample_channels = std::min(src_channels, dst_channels);
// Prepare buffers as needed for intermediate stages.
if (dst_channels < src_channels)
@ -66,8 +69,11 @@ void AudioConverter::Convert(const float* const* src,
int dst_channels,
int dst_frames,
float* const* dst) {
DCHECK(dst_channels == src_channels || dst_channels == 1 ||
src_channels == 1);
DCHECK_EQ(src_channels_, src_channels);
DCHECK_EQ(src_frames_, src_frames);
DCHECK_EQ(dst_channels_, dst_channels);
DCHECK_EQ(dst_frames_, dst_frames);;
if (src_channels == dst_channels && src_frames == dst_frames) {
// Shortcut copy.
if (src != dst) {

4
webrtc/common_audio/audio_converter.h
View File

@ -40,6 +40,10 @@ class AudioConverter {
float* const* dest);
private:
const int src_channels_;
const int src_frames_;
const int dst_channels_;
const int dst_frames_;
scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
ScopedVector<PushSincResampler> resamplers_;