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Support arbitrary input/output rates and downmixing in AudioProcessing.

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Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-22 21:00:04 +00:00
parent 34fe0153b9
commit ddbb8a2c24
23 changed files with 1291 additions and 693 deletions
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39
webrtc/modules/audio_processing/test/test_utils.h
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@ -9,6 +9,7 @@
*/
#include "webrtc/audio_processing/debug.pb.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@ -18,37 +19,6 @@ namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
static const int kChunkSizeMs = 10;
// Helper to encapsulate a contiguous data buffer with access to a pointer
// array of the deinterleaved channels.
template <typename T>
class ChannelBuffer {
public:
ChannelBuffer(int samples_per_channel, int num_channels)
: data_(new T[samples_per_channel * num_channels]),
channels_(new T*[num_channels]),
samples_per_channel_(samples_per_channel) {
memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
for (int i = 0; i < num_channels; ++i)
channels_[i] = &data_[i * samples_per_channel];
}
~ChannelBuffer() {}
void CopyFrom(const void* channel_ptr, int index) {
memcpy(channels_[index], channel_ptr, samples_per_channel_ * sizeof(T));
}
T* data() { return data_.get(); }
T* channel(int index) { return channels_[index]; }
T** channels() { return channels_.get(); }
private:
scoped_ptr<T[]> data_;
scoped_ptr<T*[]> channels_;
int samples_per_channel_;
};
// Exits on failure; do not use in unit tests.
static inline FILE* OpenFile(const std::string& filename, const char* mode) {
FILE* file = fopen(filename.c_str(), mode);
@ -59,10 +29,15 @@ static inline FILE* OpenFile(const std::string& filename, const char* mode) {
return file;
}
static inline int SamplesFromRate(int rate) {
return AudioProcessing::kChunkSizeMs * rate / 1000;
}
static inline void SetFrameSampleRate(AudioFrame* frame,
int sample_rate_hz) {
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = kChunkSizeMs * sample_rate_hz / 1000;
frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
sample_rate_hz / 1000;
}
template <typename T>