zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Make ChannelBuffer aware of frequency bands

Browse Source 
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.

R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36999004

Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2015-02-10 22:52:15 +00:00
parent d7472b52d6
commit d35a5c3506
23 changed files with 430 additions and 432 deletions
Download Patch File Download Diff File Expand all files Collapse all files

68
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
View File

@ -62,15 +62,17 @@ const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
sizeof(*kProcessSampleRates);
void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
ChannelBuffer<int16_t> cb_int(cb->samples_per_channel(),
ChannelBuffer<int16_t> cb_int(cb->num_frames(),
cb->num_channels());
Deinterleave(int_data,
cb->samples_per_channel(),
cb->num_frames(),
cb->num_channels(),
cb_int.channels());
S16ToFloat(cb_int.data(),
cb->samples_per_channel() * cb->num_channels(),
cb->data());
for (int i = 0; i < cb->num_channels(); ++i) {
S16ToFloat(cb_int.channels()[i],
cb->num_frames(),
cb->channels()[i]);
}
}
void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
@ -294,7 +296,7 @@ void OpenFileAndReadMessage(const std::string filename,
bool ReadChunk(FILE* file, int16_t* int_data, float* float_data,
ChannelBuffer<float>* cb) {
// The files always contain stereo audio.
size_t frame_size = cb->samples_per_channel() * 2;
size_t frame_size = cb->num_frames() * 2;
size_t read_count = fread(int_data, sizeof(int16_t), frame_size, file);
if (read_count != frame_size) {
// Check that the file really ended.
@ -304,9 +306,9 @@ bool ReadChunk(FILE* file, int16_t* int_data, float* float_data,
S16ToFloat(int_data, frame_size, float_data);
if (cb->num_channels() == 1) {
MixStereoToMono(float_data, cb->data(), cb->samples_per_channel());
MixStereoToMono(float_data, cb->channels()[0], cb->num_frames());
} else {
Deinterleave(float_data, cb->samples_per_channel(), 2,
Deinterleave(float_data, cb->num_frames(), 2,
cb->channels());
}
@ -1250,12 +1252,14 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
int_data.get(),
float_data.get(),
&src_buf));
for (int j = 0; j < kNumInputChannels * kSamplesPerChannel; ++j) {
src_buf.data()[j] *= kScaleFactor;
for (int j = 0; j < kNumInputChannels; ++j) {
for (int k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
}
EXPECT_EQ(kNoErr,
apm->ProcessStream(src_buf.channels(),
src_buf.samples_per_channel(),
src_buf.num_frames(),
kSampleRateHz,
LayoutFromChannels(src_buf.num_channels()),
kSampleRateHz,
@ -1273,12 +1277,14 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
int_data.get(),
float_data.get(),
&src_buf));
for (int j = 0; j < kNumInputChannels * kSamplesPerChannel; ++j) {
src_buf.data()[j] *= kScaleFactor;
for (int j = 0; j < kNumInputChannels; ++j) {
for (int k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
}
EXPECT_EQ(kNoErr,
apm->ProcessStream(src_buf.channels(),
src_buf.samples_per_channel(),
src_buf.num_frames(),
kSampleRateHz,
LayoutFromChannels(src_buf.num_channels()),
kSampleRateHz,
@ -1648,7 +1654,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
if (msg.channel_size() > 0) {
ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
for (int i = 0; i < msg.channel_size(); ++i) {
memcpy(revfloat_cb_->channel(i), msg.channel(i).data(),
memcpy(revfloat_cb_->channels()[i],
msg.channel(i).data(),
msg.channel(i).size());
}
} else {
@ -1677,7 +1684,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
if (msg.input_channel_size() > 0) {
ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
for (int i = 0; i < msg.input_channel_size(); ++i) {
memcpy(float_cb_->channel(i), msg.input_channel(i).data(),
memcpy(float_cb_->channels()[i],
msg.input_channel(i).data(),
msg.input_channel(i).size());
}
} else {
@ -1835,7 +1843,6 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
const int num_output_channels = test->num_output_channels();
const int samples_per_channel = test->sample_rate() *
AudioProcessing::kChunkSizeMs / 1000;
const int output_length = samples_per_channel * num_output_channels;
Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
num_input_channels, num_output_channels, num_render_channels, true);
@ -1876,11 +1883,13 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
test->sample_rate(),
LayoutFromChannels(num_output_channels),
float_cb_->channels()));
FloatToS16(float_cb_->data(), output_length, output_cb.data());
for (int j = 0; j < num_output_channels; ++j) {
FloatToS16(float_cb_->channels()[j],
samples_per_channel,
output_cb.channels()[j]);
float variance = 0;
float snr = ComputeSNR(output_int16.channel(j), output_cb.channel(j),
float snr = ComputeSNR(output_int16.channels()[j],
output_cb.channels()[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.
@ -2171,7 +2180,7 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
for (int j = 0; j < 10; ++j) {
EXPECT_NOERR(ap->ProcessStream(
in_cb.channels(),
in_cb.samples_per_channel(),
in_cb.num_frames(),
in_rate,
cf[i].in_layout,
out_rate,
@ -2313,9 +2322,9 @@ class AudioProcessingTest
// Temporary buffers.
const int max_length =
2 * std::max(out_cb.samples_per_channel(),
std::max(fwd_cb.samples_per_channel(),
rev_cb.samples_per_channel()));
2 * std::max(out_cb.num_frames(),
std::max(fwd_cb.num_frames(),
rev_cb.num_frames()));
scoped_ptr<float[]> float_data(new float[max_length]);
scoped_ptr<int16_t[]> int_data(new int16_t[max_length]);
@ -2324,7 +2333,7 @@ class AudioProcessingTest
ReadChunk(near_file, int_data.get(), float_data.get(), &fwd_cb)) {
EXPECT_NOERR(ap->AnalyzeReverseStream(
rev_cb.channels(),
rev_cb.samples_per_channel(),
rev_cb.num_frames(),
reverse_rate,
LayoutFromChannels(num_reverse_channels)));
@ -2334,7 +2343,7 @@ class AudioProcessingTest
EXPECT_NOERR(ap->ProcessStream(
fwd_cb.channels(),
fwd_cb.samples_per_channel(),
fwd_cb.num_frames(),
input_rate,
LayoutFromChannels(num_input_channels),
output_rate,
@ -2342,13 +2351,14 @@ class AudioProcessingTest
out_cb.channels()));
Interleave(out_cb.channels(),
out_cb.samples_per_channel(),
out_cb.num_frames(),
out_cb.num_channels(),
float_data.get());
// Dump output to file.
ASSERT_EQ(static_cast<size_t>(out_cb.length()),
int out_length = out_cb.num_channels() * out_cb.num_frames();
ASSERT_EQ(static_cast<size_t>(out_length),
fwrite(float_data.get(), sizeof(float_data[0]),
out_cb.length(), out_file));
out_length, out_file));
analog_level = ap->gain_control()->stream_analog_level();
}