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Fix simulated analog gain in audioproc.

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* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.

BUG=1260

Review URL: https://webrtc-codereview.appspot.com/1027007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2013-01-11 23:11:29 +00:00
parent f908011eb4
commit bafdae3cfc
1 changed files with 9 additions and 19 deletions
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28
webrtc/modules/audio_processing/test/process_test.cc
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@ -136,18 +136,18 @@ void usage() {
printf(" --debug_file FILE Dump a debug recording.\n");
}
static double MicLevel2Gain(int level) {
return pow(10.0, ((level - 127.0) / 128.0 * 80.) / 20.);
static float MicLevel2Gain(int level) {
return pow(10.0f, ((level - 127.0f) / 128.0f * 40.0f) / 20.0f);
}
static void SimulateMic(int mic_level, AudioFrame* frame) {
mic_level = std::min(std::max(mic_level, 0), 255);
double mic_gain = MicLevel2Gain(mic_level);
float mic_gain = MicLevel2Gain(mic_level);
int num_samples = frame->samples_per_channel_ * frame->num_channels_;
double v;
float v;
for (int n = 0; n < num_samples; n++) {
v = floor(frame->data_[n] * mic_gain + 0.5);
v = std::max(std::min(32767., v), -32768.);
v = std::max(std::min(32767.0f, v), -32768.0f);
frame->data_[n] = static_cast<int16_t>(v);
}
}
@ -618,6 +618,7 @@ void void_main(int argc, char* argv[]) {
far_frame.num_channels_ = msg.num_reverse_channels();
near_frame.sample_rate_hz_ = msg.sample_rate();
near_frame.samples_per_channel_ = samples_per_channel;
near_frame.num_channels_ = msg.num_input_channels();
if (verbose) {
printf("Init at frame: %d (primary), %d (reverse)\n",
@ -680,10 +681,6 @@ void void_main(int argc, char* argv[]) {
fflush(stdout);
}
if (apm->gain_control()->mode() == GainControl::kAdaptiveAnalog) {
SimulateMic(capture_level, &near_frame);
}
if (perf_testing) {
t0 = TickTime::Now();
}
@ -703,8 +700,6 @@ void void_main(int argc, char* argv[]) {
err == apm->kBadStreamParameterWarning);
ASSERT_TRUE(near_frame.num_channels_ == apm->num_output_channels());
capture_level = apm->gain_control()->stream_analog_level();
stream_has_voice =
static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
if (vad_out_file != NULL) {
@ -722,10 +717,6 @@ void void_main(int argc, char* argv[]) {
ns_prob_file));
}
if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
ASSERT_EQ(msg.level(), capture_level);
}
if (perf_testing) {
t1 = TickTime::Now();
TickInterval tick_diff = t1 - t0;
@ -888,7 +879,8 @@ void void_main(int argc, char* argv[]) {
fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
}
if (apm->gain_control()->mode() == GainControl::kAdaptiveAnalog) {
if (apm->gain_control()->is_enabled() &&
apm->gain_control()->mode() == GainControl::kAdaptiveAnalog) {
SimulateMic(capture_level, &near_frame);
}
@ -896,9 +888,7 @@ void void_main(int argc, char* argv[]) {
t0 = TickTime::Now();
}
// TODO(ajm): fake an analog gain while simulating.
int capture_level_in = capture_level;
const int capture_level_in = capture_level;
ASSERT_EQ(apm->kNoError,
apm->gain_control()->set_stream_analog_level(capture_level));
ASSERT_EQ(apm->kNoError,