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Increase the maximum supported sample rate to 384000 Hz and add tests

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This CL increases the maximum supported sample rate so that all rates
up to 384000 Hz are handled.

The CL also adds tests that verifies that APM works as intended for
different combinations of number of channels and sample rates.

Bug: webrtc:10882
Change-Id: I98738e33ac21413ae00fec10bb43b8796ae2078c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150532
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29014}
This commit is contained in:
Per Åhgren 2019-08-29 23:38:40 +02:00 committed by Commit Bot
parent 8577729a01
commit 3e8bf282c4
3 changed files with 151 additions and 3 deletions
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5
modules/audio_processing/audio_buffer.cc
View File

@ -25,8 +25,7 @@ namespace {
constexpr size_t kSamplesPer32kHzChannel = 320;
constexpr size_t kSamplesPer48kHzChannel = 480;
constexpr size_t kSamplesPer192kHzChannel = 1920;
constexpr size_t kMaxSamplesPerChannel = kSamplesPer192kHzChannel;
constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;
size_t NumBandsFromFramesPerChannel(size_t num_frames) {
if (num_frames == kSamplesPer32kHzChannel) {
@ -123,7 +122,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
const bool resampling_needed = input_num_frames_ != buffer_num_frames_;
if (downmix_needed) {
RTC_DCHECK_GT(kMaxSamplesPerChannel, input_num_frames_);
RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);
std::array<float, kMaxSamplesPerChannel> downmix;
if (downmix_by_averaging_) {

1
modules/audio_processing/audio_buffer.h
View File

@ -33,6 +33,7 @@ enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
class AudioBuffer {
public:
static const int kSplitBandSize = 160;
static const size_t kMaxSampleRate = 384000;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,

148
modules/audio_processing/audio_processing_unittest.cc
View File

@ -2257,6 +2257,128 @@ INSTANTIATE_TEST_SUITE_P(
std::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
#endif
// Produces a scoped trace debug output.
std::string ProduceDebugText(int render_input_sample_rate_hz,
int render_output_sample_rate_hz,
int capture_input_sample_rate_hz,
int capture_output_sample_rate_hz,
size_t render_input_num_channels,
size_t render_output_num_channels,
size_t capture_input_num_channels,
size_t capture_output_num_channels) {
rtc::StringBuilder ss;
ss << "Sample rates:"
<< "\n"
<< " Render input: " << render_input_sample_rate_hz << " Hz"
<< "\n"
<< " Render output: " << render_output_sample_rate_hz << " Hz"
<< "\n"
<< " Capture input: " << capture_input_sample_rate_hz << " Hz"
<< "\n"
<< " Capture output: " << capture_output_sample_rate_hz << " Hz"
<< "\n"
<< "Number of channels:"
<< "\n"
<< " Render input: " << render_input_num_channels << "\n"
<< " Render output: " << render_output_num_channels << "\n"
<< " Capture input: " << capture_input_num_channels << "\n"
<< " Capture output: " << capture_output_num_channels;
return ss.Release();
}
// Validates that running the audio processing module using various combinations
// of sample rates and number of channels works as intended.
void RunApmRateAndChannelTest(
rtc::ArrayView<const int> sample_rates_hz,
rtc::ArrayView<const int> render_channel_counts,
rtc::ArrayView<const int> capture_channel_counts) {
std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
webrtc::AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm->ApplyConfig(apm_config);
StreamConfig render_input_stream_config;
StreamConfig render_output_stream_config;
StreamConfig capture_input_stream_config;
StreamConfig capture_output_stream_config;
std::vector<float> render_input_frame_channels;
std::vector<float*> render_input_frame;
std::vector<float> render_output_frame_channels;
std::vector<float*> render_output_frame;
std::vector<float> capture_input_frame_channels;
std::vector<float*> capture_input_frame;
std::vector<float> capture_output_frame_channels;
std::vector<float*> capture_output_frame;
for (auto render_input_sample_rate_hz : sample_rates_hz) {
for (auto render_output_sample_rate_hz : sample_rates_hz) {
for (auto capture_input_sample_rate_hz : sample_rates_hz) {
for (auto capture_output_sample_rate_hz : sample_rates_hz) {
for (size_t render_input_num_channels : render_channel_counts) {
for (size_t capture_input_num_channels : capture_channel_counts) {
size_t render_output_num_channels = render_input_num_channels;
size_t capture_output_num_channels = capture_input_num_channels;
auto populate_audio_frame = [](int sample_rate_hz,
size_t num_channels,
StreamConfig* cfg,
std::vector<float>* channels_data,
std::vector<float*>* frame_data) {
cfg->set_sample_rate_hz(sample_rate_hz);
cfg->set_num_channels(num_channels);
cfg->set_has_keyboard(false);
size_t max_frame_size = ceil(sample_rate_hz / 100.f);
channels_data->resize(num_channels * max_frame_size);
std::fill(channels_data->begin(), channels_data->end(), 0.5f);
frame_data->resize(num_channels);
for (size_t channel = 0; channel < num_channels; ++channel) {
(*frame_data)[channel] =
&(*channels_data)[channel * max_frame_size];
}
};
populate_audio_frame(
render_input_sample_rate_hz, render_input_num_channels,
&render_input_stream_config, &render_input_frame_channels,
&render_input_frame);
populate_audio_frame(
render_output_sample_rate_hz, render_output_num_channels,
&render_output_stream_config, &render_output_frame_channels,
&render_output_frame);
populate_audio_frame(
capture_input_sample_rate_hz, capture_input_num_channels,
&capture_input_stream_config, &capture_input_frame_channels,
&capture_input_frame);
populate_audio_frame(
capture_output_sample_rate_hz, capture_output_num_channels,
&capture_output_stream_config, &capture_output_frame_channels,
&capture_output_frame);
for (size_t frame = 0; frame < 2; ++frame) {
SCOPED_TRACE(ProduceDebugText(
render_input_sample_rate_hz, render_output_sample_rate_hz,
capture_input_sample_rate_hz, capture_output_sample_rate_hz,
render_input_num_channels, render_output_num_channels,
render_input_num_channels, capture_output_num_channels));
int result = apm->ProcessReverseStream(
&render_input_frame[0], render_input_stream_config,
render_output_stream_config, &render_output_frame[0]);
EXPECT_EQ(result, AudioProcessing::kNoError);
result = apm->ProcessStream(
&capture_input_frame[0], capture_input_stream_config,
capture_output_stream_config, &capture_output_frame[0]);
EXPECT_EQ(result, AudioProcessing::kNoError);
}
}
}
}
}
}
}
}
} // namespace
TEST(RuntimeSettingTest, TestDefaultCtor) {
@ -2622,4 +2744,30 @@ TEST(ApmStatistics, ReportHasVoice) {
EXPECT_EQ(apm->ProcessStream(&frame), 0);
EXPECT_FALSE(apm->GetStatistics(false).voice_detected);
}
TEST(ApmConfiguration, HandlingOfRateAndChannelCombinations) {
std::array<int, 3> sample_rates_hz = {16000, 32000, 48000};
std::array<int, 2> render_channel_counts = {1, 7};
std::array<int, 2> capture_channel_counts = {1, 7};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
TEST(ApmConfiguration, HandlingOfChannelCombinations) {
std::array<int, 1> sample_rates_hz = {48000};
std::array<int, 8> render_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
std::array<int, 8> capture_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
TEST(ApmConfiguration, HandlingOfRateCombinations) {
std::array<int, 9> sample_rates_hz = {8000, 11025, 16000, 22050, 32000,
48000, 96000, 192000, 384000};
std::array<int, 1> render_channel_counts = {2};
std::array<int, 1> capture_channel_counts = {2};
RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
capture_channel_counts);
}
} // namespace webrtc