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Revert part of r7561, "Refactor audio conversion functions."
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Specifically, revert this part:

  "Remove hacks in AudioBuffer intended to maintain bit-exactness with
   the float path. The conversions etc. are now all natural, and
   instead we enforce close but not bit-exact output between the two
   paths."

But keep the conversion function rename, since that doesn't seem to be
causing problems.

R=tina.legrand@webrtc.org, bjornv@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7569 4adac7df-926f-26a2-2b94-8c16560cd09d
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kwiberg@webrtc.org committed Oct 30, 2014
1 parent 8219529 commit bcfb4d0
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Showing 5 changed files with 42 additions and 60 deletions.
Binary file modified data/audio_processing/output_data_fixed.pb
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Binary file modified data/audio_processing/output_data_float.pb
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38 changes: 28 additions & 10 deletions webrtc/modules/audio_processing/audio_buffer.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -51,11 +51,18 @@ int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
return -1;
}

template <typename T>
void StereoToMono(const T* left, const T* right, T* out,
void StereoToMono(const float* left, const float* right, float* out,
int samples_per_channel) {
for (int i = 0; i < samples_per_channel; ++i)
for (int i = 0; i < samples_per_channel; ++i) {
out[i] = (left[i] + right[i]) / 2;
}
}

void StereoToMono(const int16_t* left, const int16_t* right, int16_t* out,
int samples_per_channel) {
for (int i = 0; i < samples_per_channel; ++i) {
out[i] = (left[i] + right[i]) >> 1;
}
}

} // namespace
Expand Down Expand Up @@ -107,7 +114,13 @@ class IFChannelBuffer {
void RefreshI() {
if (!ivalid_) {
assert(fvalid_);
FloatS16ToS16(fbuf_.data(), ibuf_.length(), ibuf_.data());
const float* const float_data = fbuf_.data();
int16_t* const int_data = ibuf_.data();
const int length = ibuf_.length();
for (int i = 0; i < length; ++i)
int_data[i] = WEBRTC_SPL_SAT(std::numeric_limits<int16_t>::max(),
float_data[i],
std::numeric_limits<int16_t>::min());
ivalid_ = true;
}
}
Expand Down Expand Up @@ -217,8 +230,8 @@ void AudioBuffer::CopyFrom(const float* const* data,

// Convert to int16.
for (int i = 0; i < num_proc_channels_; ++i) {
FloatToFloatS16(data_ptr[i], proc_samples_per_channel_,
channels_->fbuf()->channel(i));
FloatToS16(data_ptr[i], proc_samples_per_channel_,
channels_->ibuf()->channel(i));
}
}

Expand All @@ -235,9 +248,9 @@ void AudioBuffer::CopyTo(int samples_per_channel,
data_ptr = process_buffer_->channels();
}
for (int i = 0; i < num_proc_channels_; ++i) {
FloatS16ToFloat(channels_->fbuf()->channel(i),
proc_samples_per_channel_,
data_ptr[i]);
S16ToFloat(channels_->ibuf()->channel(i),
proc_samples_per_channel_,
data_ptr[i]);
}

// Resample.
Expand Down Expand Up @@ -436,7 +449,12 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
// Downmix directly; no explicit deinterleaving needed.
int16_t* downmixed = channels_->ibuf()->channel(0);
for (int i = 0; i < input_samples_per_channel_; ++i) {
downmixed[i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
// HACK(ajm): The downmixing in the int16_t path is in practice never
// called from production code. We do this weird scaling to and from float
// to satisfy tests checking for bit-exactness with the float path.
float downmix_float = (S16ToFloat(frame->data_[i * 2]) +
S16ToFloat(frame->data_[i * 2 + 1])) / 2;
downmixed[i] = FloatToS16(downmix_float);
}
} else {
assert(num_proc_channels_ == num_input_channels_);
Expand Down
41 changes: 14 additions & 27 deletions webrtc/modules/audio_processing/test/audio_processing_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -1650,7 +1650,7 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) {
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}

TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
TEST_F(ApmTest, FloatAndIntInterfacesGiveIdenticalResults) {
audioproc::OutputData ref_data;
OpenFileAndReadMessage(ref_filename_, &ref_data);

Expand Down Expand Up @@ -1679,8 +1679,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
Init(fapm.get());

ChannelBuffer<int16_t> output_cb(samples_per_channel, num_input_channels);
ChannelBuffer<int16_t> output_int16(samples_per_channel,
num_input_channels);
scoped_ptr<int16_t[]> output_int16(new int16_t[output_length]);

int analog_level = 127;
while (ReadFrame(far_file_, revframe_, revfloat_cb_.get()) &&
Expand All @@ -1702,9 +1701,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
EXPECT_NOERR(fapm->gain_control()->set_stream_analog_level(analog_level));

EXPECT_NOERR(apm_->ProcessStream(frame_));
Deinterleave(frame_->data_, samples_per_channel, num_output_channels,
output_int16.channels());

// TODO(ajm): Update to support different output rates.
EXPECT_NOERR(fapm->ProcessStream(
float_cb_->channels(),
samples_per_channel,
Expand All @@ -1714,34 +1711,24 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
LayoutFromChannels(num_output_channels),
float_cb_->channels()));

// Convert to interleaved int16.
FloatToS16(float_cb_->data(), output_length, output_cb.data());
for (int j = 0; j < num_output_channels; ++j) {
float variance = 0;
float snr = ComputeSNR(output_int16.channel(j), output_cb.channel(j),
samples_per_channel, &variance);
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// There are a few chunks in the fixed-point profile that give low SNR.
// Listening confirmed the difference is acceptable.
const float kVarianceThreshold = 150;
const float kSNRThreshold = 10;
#else
const float kVarianceThreshold = 20;
const float kSNRThreshold = 20;
#endif
// Skip frames with low energy.
if (sqrt(variance) > kVarianceThreshold) {
EXPECT_LT(kSNRThreshold, snr);
}
}
Interleave(output_cb.channels(),
samples_per_channel,
num_output_channels,
output_int16.get());
// Verify float and int16 paths produce identical output.
EXPECT_EQ(0, memcmp(frame_->data_, output_int16.get(), output_length));

analog_level = fapm->gain_control()->stream_analog_level();
EXPECT_EQ(apm_->gain_control()->stream_analog_level(),
fapm->gain_control()->stream_analog_level());
EXPECT_EQ(apm_->echo_cancellation()->stream_has_echo(),
fapm->echo_cancellation()->stream_has_echo());
EXPECT_NEAR(apm_->noise_suppression()->speech_probability(),
fapm->noise_suppression()->speech_probability(),
0.0005);
EXPECT_EQ(apm_->voice_detection()->stream_has_voice(),
fapm->voice_detection()->stream_has_voice());
EXPECT_EQ(apm_->noise_suppression()->speech_probability(),
fapm->noise_suppression()->speech_probability());

// Reset in case of downmixing.
frame_->num_channels_ = test->num_input_channels();
Expand Down
23 changes: 0 additions & 23 deletions webrtc/modules/audio_processing/test/test_utils.h
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Original file line number Diff line number Diff line change
Expand Up @@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/

#include <math.h>
#include <limits>

#include "webrtc/audio_processing/debug.pb.h"
Expand Down Expand Up @@ -154,26 +153,4 @@ static inline bool ReadMessageFromFile(FILE* file,
return msg->ParseFromArray(bytes.get(), size);
}

template <typename T>
float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (int i = 0; i < length; ++i) {
T error = ref[i] - test[i];
mse += error * error;
*variance += ref[i] * ref[i];
mean += ref[i];
}
mse /= length;
*variance /= length;
mean /= length;
*variance -= mean * mean;

float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(*variance / mse);
return snr;
}

} // namespace webrtc

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