diff options
Diffstat (limited to 'media/libcubeb/test/test_resampler.cpp')
-rw-r--r-- | media/libcubeb/test/test_resampler.cpp | 1164 |
1 files changed, 1164 insertions, 0 deletions
diff --git a/media/libcubeb/test/test_resampler.cpp b/media/libcubeb/test/test_resampler.cpp new file mode 100644 index 0000000000..adbcfd8c5e --- /dev/null +++ b/media/libcubeb/test/test_resampler.cpp @@ -0,0 +1,1164 @@ +/* + * Copyright © 2016 Mozilla Foundation + * + * This program is made available under an ISC-style license. See the + * accompanying file LICENSE for details. + */ +#ifndef NOMINMAX +#define NOMINMAX +#endif // NOMINMAX +#include "common.h" +#include "cubeb_resampler_internal.h" +#include "gtest/gtest.h" +#include <algorithm> +#include <iostream> +#include <stdio.h> + +/* Windows cmath USE_MATH_DEFINE thing... */ +const float PI = 3.14159265359f; + +/* Testing all sample rates is very long, so if THOROUGH_TESTING is not defined, + * only part of the test suite is ran. */ +#ifdef THOROUGH_TESTING +/* Some standard sample rates we're testing with. */ +const uint32_t sample_rates[] = {8000, 16000, 32000, 44100, + 48000, 88200, 96000, 192000}; +/* The maximum number of channels we're resampling. */ +const uint32_t max_channels = 2; +/* The minimum an maximum number of milliseconds we're resampling for. This is + * used to simulate the fact that the audio stream is resampled in chunks, + * because audio is delivered using callbacks. */ +const uint32_t min_chunks = 10; /* ms */ +const uint32_t max_chunks = 30; /* ms */ +const uint32_t chunk_increment = 1; + +#else + +const uint32_t sample_rates[] = { + 8000, + 44100, + 48000, +}; +const uint32_t max_channels = 2; +const uint32_t min_chunks = 10; /* ms */ +const uint32_t max_chunks = 30; /* ms */ +const uint32_t chunk_increment = 10; +#endif + +// #define DUMP_ARRAYS +#ifdef DUMP_ARRAYS +/** + * Files produced by dump(...) can be converted to .wave files using: + * + * sox -c <channel_count> -r <rate> -e float -b 32 file.raw file.wav + * + * for floating-point audio, or: + * + * sox -c <channel_count> -r <rate> -e unsigned -b 16 file.raw file.wav + * + * for 16bit integer audio. + */ + +/* Use the correct implementation of fopen, depending on the platform. */ +void +fopen_portable(FILE ** f, const char * name, const char * mode) +{ +#ifdef WIN32 + fopen_s(f, name, mode); +#else + *f = fopen(name, mode); +#endif +} + +template <typename T> +void +dump(const char * name, T * frames, size_t count) +{ + FILE * file; + fopen_portable(&file, name, "wb"); + + if (!file) { + fprintf(stderr, "error opening %s\n", name); + return; + } + + if (count != fwrite(frames, sizeof(T), count, file)) { + fprintf(stderr, "error writing to %s\n", name); + } + fclose(file); +} +#else +template <typename T> +void +dump(const char * name, T * frames, size_t count) +{ +} +#endif + +// The more the ratio is far from 1, the more we accept a big error. +float +epsilon_tweak_ratio(float ratio) +{ + return ratio >= 1 ? ratio : 1 / ratio; +} + +// Epsilon values for comparing resampled data to expected data. +// The bigger the resampling ratio is, the more lax we are about errors. +template <typename T> +T +epsilon(float ratio); + +template <> +float +epsilon(float ratio) +{ + return 0.08f * epsilon_tweak_ratio(ratio); +} + +template <> +int16_t +epsilon(float ratio) +{ + return static_cast<int16_t>(10 * epsilon_tweak_ratio(ratio)); +} + +void +test_delay_lines(uint32_t delay_frames, uint32_t channels, uint32_t chunk_ms) +{ + const size_t length_s = 2; + const size_t rate = 44100; + const size_t length_frames = rate * length_s; + delay_line<float> delay(delay_frames, channels, rate); + auto_array<float> input; + auto_array<float> output; + uint32_t chunk_length = channels * chunk_ms * rate / 1000; + uint32_t output_offset = 0; + uint32_t channel = 0; + + /** Generate diracs every 100 frames, and check they are delayed. */ + input.push_silence(length_frames * channels); + for (uint32_t i = 0; i < input.length() - 1; i += 100) { + input.data()[i + channel] = 0.5; + channel = (channel + 1) % channels; + } + dump("input.raw", input.data(), input.length()); + while (input.length()) { + uint32_t to_pop = + std::min<uint32_t>(input.length(), chunk_length * channels); + float * in = delay.input_buffer(to_pop / channels); + input.pop(in, to_pop); + delay.written(to_pop / channels); + output.push_silence(to_pop); + delay.output(output.data() + output_offset, to_pop / channels); + output_offset += to_pop; + } + + // Check the diracs have been shifted by `delay_frames` frames. + for (uint32_t i = 0; i < output.length() - delay_frames * channels + 1; + i += 100) { + ASSERT_EQ(output.data()[i + channel + delay_frames * channels], 0.5); + channel = (channel + 1) % channels; + } + + dump("output.raw", output.data(), output.length()); +} +/** + * This takes sine waves with a certain `channels` count, `source_rate`, and + * resample them, by chunk of `chunk_duration` milliseconds, to `target_rate`. + * Then a sample-wise comparison is performed against a sine wave generated at + * the correct rate. + */ +template <typename T> +void +test_resampler_one_way(uint32_t channels, uint32_t source_rate, + uint32_t target_rate, float chunk_duration) +{ + size_t chunk_duration_in_source_frames = + static_cast<uint32_t>(ceil(chunk_duration * source_rate / 1000.)); + float resampling_ratio = static_cast<float>(source_rate) / target_rate; + cubeb_resampler_speex_one_way<T> resampler(channels, source_rate, target_rate, + 3); + auto_array<T> source(channels * source_rate * 10); + auto_array<T> destination(channels * target_rate * 10); + auto_array<T> expected(channels * target_rate * 10); + uint32_t phase_index = 0; + uint32_t offset = 0; + const uint32_t buf_len = 2; /* seconds */ + + // generate a sine wave in each channel, at the source sample rate + source.push_silence(channels * source_rate * buf_len); + while (offset != source.length()) { + float p = phase_index++ / static_cast<float>(source_rate); + for (uint32_t j = 0; j < channels; j++) { + source.data()[offset++] = 0.5 * sin(440. * 2 * PI * p); + } + } + + dump("input.raw", source.data(), source.length()); + + expected.push_silence(channels * target_rate * buf_len); + // generate a sine wave in each channel, at the target sample rate. + // Insert silent samples at the beginning to account for the resampler + // latency. + offset = resampler.latency() * channels; + for (uint32_t i = 0; i < offset; i++) { + expected.data()[i] = 0.0f; + } + phase_index = 0; + while (offset != expected.length()) { + float p = phase_index++ / static_cast<float>(target_rate); + for (uint32_t j = 0; j < channels; j++) { + expected.data()[offset++] = 0.5 * sin(440. * 2 * PI * p); + } + } + + dump("expected.raw", expected.data(), expected.length()); + + // resample by chunk + uint32_t write_offset = 0; + destination.push_silence(channels * target_rate * buf_len); + while (write_offset < destination.length()) { + size_t output_frames = static_cast<uint32_t>( + floor(chunk_duration_in_source_frames / resampling_ratio)); + uint32_t input_frames = resampler.input_needed_for_output(output_frames); + resampler.input(source.data(), input_frames); + source.pop(nullptr, input_frames * channels); + resampler.output( + destination.data() + write_offset, + std::min(output_frames, + (destination.length() - write_offset) / channels)); + write_offset += output_frames * channels; + } + + dump("output.raw", destination.data(), expected.length()); + + // compare, taking the latency into account + bool fuzzy_equal = true; + for (uint32_t i = resampler.latency() + 1; i < expected.length(); i++) { + float diff = fabs(expected.data()[i] - destination.data()[i]); + if (diff > epsilon<T>(resampling_ratio)) { + fprintf(stderr, "divergence at %d: %f %f (delta %f)\n", i, + expected.data()[i], destination.data()[i], diff); + fuzzy_equal = false; + } + } + ASSERT_TRUE(fuzzy_equal); +} + +template <typename T> +cubeb_sample_format +cubeb_format(); + +template <> +cubeb_sample_format +cubeb_format<float>() +{ + return CUBEB_SAMPLE_FLOAT32NE; +} + +template <> +cubeb_sample_format +cubeb_format<short>() +{ + return CUBEB_SAMPLE_S16NE; +} + +struct osc_state { + osc_state() + : input_phase_index(0), output_phase_index(0), output_offset(0), + input_channels(0), output_channels(0) + { + } + uint32_t input_phase_index; + uint32_t max_output_phase_index; + uint32_t output_phase_index; + uint32_t output_offset; + uint32_t input_channels; + uint32_t output_channels; + uint32_t output_rate; + uint32_t target_rate; + auto_array<float> input; + auto_array<float> output; +}; + +uint32_t +fill_with_sine(float * buf, uint32_t rate, uint32_t channels, uint32_t frames, + uint32_t initial_phase) +{ + uint32_t offset = 0; + for (uint32_t i = 0; i < frames; i++) { + float p = initial_phase++ / static_cast<float>(rate); + for (uint32_t j = 0; j < channels; j++) { + buf[offset++] = 0.5 * sin(440. * 2 * PI * p); + } + } + return initial_phase; +} + +long +data_cb_resampler(cubeb_stream * /*stm*/, void * user_ptr, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + osc_state * state = reinterpret_cast<osc_state *>(user_ptr); + const float * in = reinterpret_cast<const float *>(input_buffer); + float * out = reinterpret_cast<float *>(output_buffer); + + state->input.push(in, frame_count * state->input_channels); + + /* Check how much output frames we need to write */ + uint32_t remaining = + state->max_output_phase_index - state->output_phase_index; + uint32_t to_write = std::min<uint32_t>(remaining, frame_count); + state->output_phase_index = + fill_with_sine(out, state->target_rate, state->output_channels, to_write, + state->output_phase_index); + + return to_write; +} + +template <typename T> +bool +array_fuzzy_equal(const auto_array<T> & lhs, const auto_array<T> & rhs, T epsi) +{ + uint32_t len = std::min(lhs.length(), rhs.length()); + + for (uint32_t i = 0; i < len; i++) { + if (fabs(lhs.at(i) - rhs.at(i)) > epsi) { + std::cout << "not fuzzy equal at index: " << i << " lhs: " << lhs.at(i) + << " rhs: " << rhs.at(i) + << " delta: " << fabs(lhs.at(i) - rhs.at(i)) + << " epsilon: " << epsi << std::endl; + return false; + } + } + return true; +} + +template <typename T> +void +test_resampler_duplex(uint32_t input_channels, uint32_t output_channels, + uint32_t input_rate, uint32_t output_rate, + uint32_t target_rate, float chunk_duration) +{ + cubeb_stream_params input_params; + cubeb_stream_params output_params; + osc_state state; + + input_params.format = output_params.format = cubeb_format<T>(); + state.input_channels = input_params.channels = input_channels; + state.output_channels = output_params.channels = output_channels; + input_params.rate = input_rate; + state.output_rate = output_params.rate = output_rate; + state.target_rate = target_rate; + input_params.prefs = output_params.prefs = CUBEB_STREAM_PREF_NONE; + long got; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, &input_params, &output_params, target_rate, + data_cb_resampler, (void *)&state, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + long latency = cubeb_resampler_latency(resampler); + + const uint32_t duration_s = 2; + int32_t duration_frames = duration_s * target_rate; + uint32_t input_array_frame_count = + ceil(chunk_duration * input_rate / 1000) + + ceilf(static_cast<float>(input_rate) / target_rate) * 2; + uint32_t output_array_frame_count = chunk_duration * output_rate / 1000; + auto_array<float> input_buffer(input_channels * input_array_frame_count); + auto_array<float> output_buffer(output_channels * output_array_frame_count); + auto_array<float> expected_resampled_input(input_channels * duration_frames); + auto_array<float> expected_resampled_output(output_channels * output_rate * + duration_s); + + state.max_output_phase_index = duration_s * target_rate; + + expected_resampled_input.push_silence(input_channels * duration_frames); + expected_resampled_output.push_silence(output_channels * output_rate * + duration_s); + + /* expected output is a 440Hz sine wave at 16kHz */ + fill_with_sine(expected_resampled_input.data() + latency, target_rate, + input_channels, duration_frames - latency, 0); + /* expected output is a 440Hz sine wave at 32kHz */ + fill_with_sine(expected_resampled_output.data() + latency, output_rate, + output_channels, output_rate * duration_s - latency, 0); + + while (state.output_phase_index != state.max_output_phase_index) { + uint32_t leftover_samples = input_buffer.length() * input_channels; + input_buffer.reserve(input_array_frame_count); + state.input_phase_index = fill_with_sine( + input_buffer.data() + leftover_samples, input_rate, input_channels, + input_array_frame_count - leftover_samples, state.input_phase_index); + long input_consumed = input_array_frame_count; + input_buffer.set_length(input_array_frame_count); + + got = cubeb_resampler_fill(resampler, input_buffer.data(), &input_consumed, + output_buffer.data(), output_array_frame_count); + + /* handle leftover input */ + if (input_array_frame_count != static_cast<uint32_t>(input_consumed)) { + input_buffer.pop(nullptr, input_consumed * input_channels); + } else { + input_buffer.clear(); + } + + state.output.push(output_buffer.data(), got * state.output_channels); + } + + dump("input_expected.raw", expected_resampled_input.data(), + expected_resampled_input.length()); + dump("output_expected.raw", expected_resampled_output.data(), + expected_resampled_output.length()); + dump("input.raw", state.input.data(), state.input.length()); + dump("output.raw", state.output.data(), state.output.length()); + + // This is disabled because the latency estimation in the resampler code is + // slightly off so we can generate expected vectors. + // See https://github.com/kinetiknz/cubeb/issues/93 + // ASSERT_TRUE(array_fuzzy_equal(state.input, expected_resampled_input, + // epsilon<T>(input_rate/target_rate))); + // ASSERT_TRUE(array_fuzzy_equal(state.output, expected_resampled_output, + // epsilon<T>(output_rate/target_rate))); + + cubeb_resampler_destroy(resampler); +} + +#define array_size(x) (sizeof(x) / sizeof(x[0])) + +TEST(cubeb, resampler_one_way) +{ + /* Test one way resamplers */ + for (uint32_t channels = 1; channels <= max_channels; channels++) { + for (uint32_t source_rate = 0; source_rate < array_size(sample_rates); + source_rate++) { + for (uint32_t dest_rate = 0; dest_rate < array_size(sample_rates); + dest_rate++) { + for (uint32_t chunk_duration = min_chunks; chunk_duration < max_chunks; + chunk_duration += chunk_increment) { + fprintf(stderr, + "one_way: channels: %d, source_rate: %d, dest_rate: %d, " + "chunk_duration: %d\n", + channels, sample_rates[source_rate], sample_rates[dest_rate], + chunk_duration); + test_resampler_one_way<float>(channels, sample_rates[source_rate], + sample_rates[dest_rate], + chunk_duration); + } + } + } + } +} + +TEST(cubeb, DISABLED_resampler_duplex) +{ + for (uint32_t input_channels = 1; input_channels <= max_channels; + input_channels++) { + for (uint32_t output_channels = 1; output_channels <= max_channels; + output_channels++) { + for (uint32_t source_rate_input = 0; + source_rate_input < array_size(sample_rates); source_rate_input++) { + for (uint32_t source_rate_output = 0; + source_rate_output < array_size(sample_rates); + source_rate_output++) { + for (uint32_t dest_rate = 0; dest_rate < array_size(sample_rates); + dest_rate++) { + for (uint32_t chunk_duration = min_chunks; + chunk_duration < max_chunks; + chunk_duration += chunk_increment) { + fprintf(stderr, + "input channels:%d output_channels:%d input_rate:%d " + "output_rate:%d target_rate:%d chunk_ms:%d\n", + input_channels, output_channels, + sample_rates[source_rate_input], + sample_rates[source_rate_output], sample_rates[dest_rate], + chunk_duration); + test_resampler_duplex<float>(input_channels, output_channels, + sample_rates[source_rate_input], + sample_rates[source_rate_output], + sample_rates[dest_rate], + chunk_duration); + } + } + } + } + } + } +} + +TEST(cubeb, resampler_delay_line) +{ + for (uint32_t channel = 1; channel <= 2; channel++) { + for (uint32_t delay_frames = 4; delay_frames <= 40; + delay_frames += chunk_increment) { + for (uint32_t chunk_size = 10; chunk_size <= 30; chunk_size++) { + fprintf(stderr, "channel: %d, delay_frames: %d, chunk_size: %d\n", + channel, delay_frames, chunk_size); + test_delay_lines(delay_frames, channel, chunk_size); + } + } + } +} + +long +test_output_only_noop_data_cb(cubeb_stream * /*stm*/, void * /*user_ptr*/, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + EXPECT_TRUE(output_buffer); + EXPECT_TRUE(!input_buffer); + return frame_count; +} + +TEST(cubeb, resampler_output_only_noop) +{ + cubeb_stream_params output_params; + int target_rate; + + output_params.rate = 44100; + output_params.channels = 1; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + target_rate = output_params.rate; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, nullptr, &output_params, target_rate, + test_output_only_noop_data_cb, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + const long out_frames = 128; + float out_buffer[out_frames]; + long got; + + got = + cubeb_resampler_fill(resampler, nullptr, nullptr, out_buffer, out_frames); + + ASSERT_EQ(got, out_frames); + + cubeb_resampler_destroy(resampler); +} + +long +test_drain_data_cb(cubeb_stream * /*stm*/, void * user_ptr, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + EXPECT_TRUE(output_buffer); + EXPECT_TRUE(!input_buffer); + auto cb_count = static_cast<int *>(user_ptr); + (*cb_count)++; + return frame_count - 1; +} + +TEST(cubeb, resampler_drain) +{ + cubeb_stream_params output_params; + int target_rate; + + output_params.rate = 44100; + output_params.channels = 1; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + target_rate = 48000; + int cb_count = 0; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, nullptr, &output_params, target_rate, + test_drain_data_cb, &cb_count, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + const long out_frames = 128; + float out_buffer[out_frames]; + long got; + + do { + got = cubeb_resampler_fill(resampler, nullptr, nullptr, out_buffer, + out_frames); + } while (got == out_frames); + + /* The callback should be called once but not again after returning < + * frame_count. */ + ASSERT_EQ(cb_count, 1); + + cubeb_resampler_destroy(resampler); +} + +// gtest does not support using ASSERT_EQ and friend in a function that returns +// a value. +void +check_output(const void * input_buffer, void * output_buffer, long frame_count) +{ + ASSERT_EQ(input_buffer, nullptr); + ASSERT_EQ(frame_count, 256); + ASSERT_TRUE(!!output_buffer); +} + +long +cb_passthrough_resampler_output(cubeb_stream * /*stm*/, void * /*user_ptr*/, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + check_output(input_buffer, output_buffer, frame_count); + return frame_count; +} + +TEST(cubeb, resampler_passthrough_output_only) +{ + // Test that the passthrough resampler works when there is only an output + // stream. + cubeb_stream_params output_params; + + const size_t output_channels = 2; + output_params.channels = output_channels; + output_params.rate = 44100; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + int target_rate = output_params.rate; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, nullptr, &output_params, target_rate, + cb_passthrough_resampler_output, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + float output_buffer[output_channels * 256]; + + long got; + for (uint32_t i = 0; i < 30; i++) { + got = cubeb_resampler_fill(resampler, nullptr, nullptr, output_buffer, 256); + ASSERT_EQ(got, 256); + } + + cubeb_resampler_destroy(resampler); +} + +// gtest does not support using ASSERT_EQ and friend in a function that returns +// a value. +void +check_input(const void * input_buffer, void * output_buffer, long frame_count) +{ + ASSERT_EQ(output_buffer, nullptr); + ASSERT_EQ(frame_count, 256); + ASSERT_TRUE(!!input_buffer); +} + +long +cb_passthrough_resampler_input(cubeb_stream * /*stm*/, void * /*user_ptr*/, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + check_input(input_buffer, output_buffer, frame_count); + return frame_count; +} + +TEST(cubeb, resampler_passthrough_input_only) +{ + // Test that the passthrough resampler works when there is only an output + // stream. + cubeb_stream_params input_params; + + const size_t input_channels = 2; + input_params.channels = input_channels; + input_params.rate = 44100; + input_params.format = CUBEB_SAMPLE_FLOAT32NE; + int target_rate = input_params.rate; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, &input_params, nullptr, target_rate, + cb_passthrough_resampler_input, nullptr, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + float input_buffer[input_channels * 256]; + + long got; + for (uint32_t i = 0; i < 30; i++) { + long int frames = 256; + got = cubeb_resampler_fill(resampler, input_buffer, &frames, nullptr, 0); + ASSERT_EQ(got, 256); + } + + cubeb_resampler_destroy(resampler); +} + +template <typename T> +long +seq(T * array, int stride, long start, long count) +{ + uint32_t output_idx = 0; + for (int i = 0; i < count; i++) { + for (int j = 0; j < stride; j++) { + array[output_idx + j] = static_cast<T>(start + i); + } + output_idx += stride; + } + return start + count; +} + +template <typename T> +void +is_seq(T * array, int stride, long count, long expected_start) +{ + uint32_t output_index = 0; + for (long i = 0; i < count; i++) { + for (int j = 0; j < stride; j++) { + ASSERT_EQ(array[output_index + j], expected_start + i); + } + output_index += stride; + } +} + +template <typename T> +void +is_not_seq(T * array, int stride, long count, long expected_start) +{ + uint32_t output_index = 0; + for (long i = 0; i < count; i++) { + for (int j = 0; j < stride; j++) { + ASSERT_NE(array[output_index + j], expected_start + i); + } + output_index += stride; + } +} + +struct closure { + int input_channel_count; +}; + +// gtest does not support using ASSERT_EQ and friend in a function that returns +// a value. +template <typename T> +void +check_duplex(const T * input_buffer, T * output_buffer, long frame_count, + int input_channel_count) +{ + ASSERT_EQ(frame_count, 256); + // Silence scan-build warning. + ASSERT_TRUE(!!output_buffer); + assert(output_buffer); + ASSERT_TRUE(!!input_buffer); + assert(input_buffer); + + int output_index = 0; + int input_index = 0; + for (int i = 0; i < frame_count; i++) { + // output is two channels, input one or two channels. + if (input_channel_count == 1) { + output_buffer[output_index] = output_buffer[output_index + 1] = + input_buffer[i]; + } else if (input_channel_count == 2) { + output_buffer[output_index] = input_buffer[input_index]; + output_buffer[output_index + 1] = input_buffer[input_index + 1]; + } + output_index += 2; + input_index += input_channel_count; + } +} + +long +cb_passthrough_resampler_duplex(cubeb_stream * /*stm*/, void * user_ptr, + const void * input_buffer, void * output_buffer, + long frame_count) +{ + closure * c = reinterpret_cast<closure *>(user_ptr); + check_duplex<float>(static_cast<const float *>(input_buffer), + static_cast<float *>(output_buffer), frame_count, + c->input_channel_count); + return frame_count; +} + +TEST(cubeb, resampler_passthrough_duplex_callback_reordering) +{ + // Test that when pre-buffering on resampler creation, we can survive an input + // callback being delayed. + + cubeb_stream_params input_params; + cubeb_stream_params output_params; + + const int input_channels = 1; + const int output_channels = 2; + + input_params.channels = input_channels; + input_params.rate = 44100; + input_params.format = CUBEB_SAMPLE_FLOAT32NE; + + output_params.channels = output_channels; + output_params.rate = input_params.rate; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + + int target_rate = input_params.rate; + + closure c; + c.input_channel_count = input_channels; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, &input_params, &output_params, target_rate, + cb_passthrough_resampler_duplex, &c, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + const long BUF_BASE_SIZE = 256; + float input_buffer_prebuffer[input_channels * BUF_BASE_SIZE * 2]; + float input_buffer_glitch[input_channels * BUF_BASE_SIZE * 2]; + float input_buffer_normal[input_channels * BUF_BASE_SIZE]; + float output_buffer[output_channels * BUF_BASE_SIZE]; + + long seq_idx = 0; + long output_seq_idx = 0; + + long prebuffer_frames = + ARRAY_LENGTH(input_buffer_prebuffer) / input_params.channels; + seq_idx = + seq(input_buffer_prebuffer, input_channels, seq_idx, prebuffer_frames); + + long got = + cubeb_resampler_fill(resampler, input_buffer_prebuffer, &prebuffer_frames, + output_buffer, BUF_BASE_SIZE); + + output_seq_idx += BUF_BASE_SIZE; + + // prebuffer_frames will hold the frames used by the resampler. + ASSERT_EQ(prebuffer_frames, BUF_BASE_SIZE); + ASSERT_EQ(got, BUF_BASE_SIZE); + + for (uint32_t i = 0; i < 300; i++) { + long int frames = BUF_BASE_SIZE; + // Simulate that sometimes, we don't have the input callback on time + if (i != 0 && (i % 100) == 0) { + long zero = 0; + got = + cubeb_resampler_fill(resampler, input_buffer_normal /* unused here */, + &zero, output_buffer, BUF_BASE_SIZE); + is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx); + output_seq_idx += BUF_BASE_SIZE; + } else if (i != 0 && (i % 100) == 1) { + // if this is the case, the on the next iteration, we'll have twice the + // amount of input frames + seq_idx = + seq(input_buffer_glitch, input_channels, seq_idx, BUF_BASE_SIZE * 2); + frames = 2 * BUF_BASE_SIZE; + got = cubeb_resampler_fill(resampler, input_buffer_glitch, &frames, + output_buffer, BUF_BASE_SIZE); + is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx); + output_seq_idx += BUF_BASE_SIZE; + } else { + // normal case + seq_idx = + seq(input_buffer_normal, input_channels, seq_idx, BUF_BASE_SIZE); + long normal_input_frame_count = 256; + got = cubeb_resampler_fill(resampler, input_buffer_normal, + &normal_input_frame_count, output_buffer, + BUF_BASE_SIZE); + is_seq(output_buffer, 2, BUF_BASE_SIZE, output_seq_idx); + output_seq_idx += BUF_BASE_SIZE; + } + ASSERT_EQ(got, BUF_BASE_SIZE); + } + + cubeb_resampler_destroy(resampler); +} + +// Artificially simulate output thread underruns, +// by building up artificial delay in the input. +// Check that the frame drop logic kicks in. +TEST(cubeb, resampler_drift_drop_data) +{ + for (uint32_t input_channels = 1; input_channels < 3; input_channels++) { + cubeb_stream_params input_params; + cubeb_stream_params output_params; + + const int output_channels = 2; + const int sample_rate = 44100; + + input_params.channels = input_channels; + input_params.rate = sample_rate; + input_params.format = CUBEB_SAMPLE_FLOAT32NE; + + output_params.channels = output_channels; + output_params.rate = sample_rate; + output_params.format = CUBEB_SAMPLE_FLOAT32NE; + + int target_rate = input_params.rate; + + closure c; + c.input_channel_count = input_channels; + + cubeb_resampler * resampler = cubeb_resampler_create( + (cubeb_stream *)nullptr, &input_params, &output_params, target_rate, + cb_passthrough_resampler_duplex, &c, CUBEB_RESAMPLER_QUALITY_VOIP, + CUBEB_RESAMPLER_RECLOCK_NONE); + + const long BUF_BASE_SIZE = 256; + + // The factor by which the deadline is missed. This is intentionally + // kind of large to trigger the frame drop quickly. In real life, multiple + // smaller under-runs would accumulate. + const long UNDERRUN_FACTOR = 10; + // Number buffer used for pre-buffering, that some backends do. + const long PREBUFFER_FACTOR = 2; + + std::vector<float> input_buffer_prebuffer(input_channels * BUF_BASE_SIZE * + PREBUFFER_FACTOR); + std::vector<float> input_buffer_glitch(input_channels * BUF_BASE_SIZE * + UNDERRUN_FACTOR); + std::vector<float> input_buffer_normal(input_channels * BUF_BASE_SIZE); + std::vector<float> output_buffer(output_channels * BUF_BASE_SIZE); + + long seq_idx = 0; + long output_seq_idx = 0; + + long prebuffer_frames = + input_buffer_prebuffer.size() / input_params.channels; + seq_idx = seq(input_buffer_prebuffer.data(), input_channels, seq_idx, + prebuffer_frames); + + long got = cubeb_resampler_fill(resampler, input_buffer_prebuffer.data(), + &prebuffer_frames, output_buffer.data(), + BUF_BASE_SIZE); + + output_seq_idx += BUF_BASE_SIZE; + + // prebuffer_frames will hold the frames used by the resampler. + ASSERT_EQ(prebuffer_frames, BUF_BASE_SIZE); + ASSERT_EQ(got, BUF_BASE_SIZE); + + for (uint32_t i = 0; i < 300; i++) { + long int frames = BUF_BASE_SIZE; + if (i != 0 && (i % 100) == 1) { + // Once in a while, the output thread misses its deadline. + // The input thread still produces data, so it ends up accumulating. + // Simulate this by providing a much bigger input buffer. Check that the + // sequence is now unaligned, meaning we've dropped data to keep + // everything in sync. + seq_idx = seq(input_buffer_glitch.data(), input_channels, seq_idx, + BUF_BASE_SIZE * UNDERRUN_FACTOR); + frames = BUF_BASE_SIZE * UNDERRUN_FACTOR; + got = + cubeb_resampler_fill(resampler, input_buffer_glitch.data(), &frames, + output_buffer.data(), BUF_BASE_SIZE); + is_seq(output_buffer.data(), 2, BUF_BASE_SIZE, output_seq_idx); + output_seq_idx += BUF_BASE_SIZE; + } else if (i != 0 && (i % 100) == 2) { + // On the next iteration, the sequence should be broken + seq_idx = seq(input_buffer_normal.data(), input_channels, seq_idx, + BUF_BASE_SIZE); + long normal_input_frame_count = 256; + got = cubeb_resampler_fill(resampler, input_buffer_normal.data(), + &normal_input_frame_count, + output_buffer.data(), BUF_BASE_SIZE); + is_not_seq(output_buffer.data(), output_channels, BUF_BASE_SIZE, + output_seq_idx); + // Reclock so that we can use is_seq again. + output_seq_idx = output_buffer[BUF_BASE_SIZE * output_channels - 1] + 1; + } else { + // normal case + seq_idx = seq(input_buffer_normal.data(), input_channels, seq_idx, + BUF_BASE_SIZE); + long normal_input_frame_count = 256; + got = cubeb_resampler_fill(resampler, input_buffer_normal.data(), + &normal_input_frame_count, + output_buffer.data(), BUF_BASE_SIZE); + is_seq(output_buffer.data(), output_channels, BUF_BASE_SIZE, + output_seq_idx); + output_seq_idx += BUF_BASE_SIZE; + } + ASSERT_EQ(got, BUF_BASE_SIZE); + } + + cubeb_resampler_destroy(resampler); + } +} + +static long +passthrough_resampler_fill_eq_input(cubeb_stream * stream, void * user_ptr, + void const * input_buffer, + void * output_buffer, long nframes) +{ + // gtest does not support using ASSERT_EQ and friends in a + // function that returns a value. + [nframes, input_buffer]() { + ASSERT_EQ(nframes, 32); + const float * input = static_cast<const float *>(input_buffer); + for (int i = 0; i < 64; ++i) { + ASSERT_FLOAT_EQ(input[i], 0.01 * i); + } + }(); + return nframes; +} + +TEST(cubeb, passthrough_resampler_fill_eq_input) +{ + uint32_t channels = 2; + uint32_t sample_rate = 44100; + passthrough_resampler<float> resampler = + passthrough_resampler<float>(nullptr, passthrough_resampler_fill_eq_input, + nullptr, channels, sample_rate); + + long input_frame_count = 32; + long output_frame_count = 32; + float input[64] = {}; + float output[64] = {}; + for (uint32_t i = 0; i < input_frame_count * channels; ++i) { + input[i] = 0.01 * i; + } + long got = + resampler.fill(input, &input_frame_count, output, output_frame_count); + ASSERT_EQ(got, output_frame_count); + // Input frames used must be equal to output frames. + ASSERT_EQ(input_frame_count, output_frame_count); +} + +static long +passthrough_resampler_fill_short_input(cubeb_stream * stream, void * user_ptr, + void const * input_buffer, + void * output_buffer, long nframes) +{ + // gtest does not support using ASSERT_EQ and friends in a + // function that returns a value. + [nframes, input_buffer]() { + ASSERT_EQ(nframes, 32); + const float * input = static_cast<const float *>(input_buffer); + // First part contains the input + for (int i = 0; i < 32; ++i) { + ASSERT_FLOAT_EQ(input[i], 0.01 * i); + } + // missing part contains silence + for (int i = 32; i < 64; ++i) { + ASSERT_FLOAT_EQ(input[i], 0.0); + } + }(); + return nframes; +} + +TEST(cubeb, passthrough_resampler_fill_short_input) +{ + uint32_t channels = 2; + uint32_t sample_rate = 44100; + passthrough_resampler<float> resampler = passthrough_resampler<float>( + nullptr, passthrough_resampler_fill_short_input, nullptr, channels, + sample_rate); + + long input_frame_count = 16; + long output_frame_count = 32; + float input[64] = {}; + float output[64] = {}; + for (uint32_t i = 0; i < input_frame_count * channels; ++i) { + input[i] = 0.01 * i; + } + long got = + resampler.fill(input, &input_frame_count, output, output_frame_count); + ASSERT_EQ(got, output_frame_count); + // Input frames used are less than the output frames due to glitch. + ASSERT_EQ(input_frame_count, output_frame_count - 16); +} + +static long +passthrough_resampler_fill_input_left(cubeb_stream * stream, void * user_ptr, + void const * input_buffer, + void * output_buffer, long nframes) +{ + // gtest does not support using ASSERT_EQ and friends in a + // function that returns a value. + int iteration = *static_cast<int *>(user_ptr); + if (iteration == 1) { + [nframes, input_buffer]() { + ASSERT_EQ(nframes, 32); + const float * input = static_cast<const float *>(input_buffer); + for (int i = 0; i < 64; ++i) { + ASSERT_FLOAT_EQ(input[i], 0.01 * i); + } + }(); + } else if (iteration == 2) { + [nframes, input_buffer]() { + ASSERT_EQ(nframes, 32); + const float * input = static_cast<const float *>(input_buffer); + for (int i = 0; i < 32; ++i) { + // First part contains the reamaining input samples from previous + // iteration (since they were more). + ASSERT_FLOAT_EQ(input[i], 0.01 * (i + 64)); + // next part contains the new buffer + ASSERT_FLOAT_EQ(input[i + 32], 0.01 * i); + } + }(); + } else if (iteration == 3) { + [nframes, input_buffer]() { + ASSERT_EQ(nframes, 32); + const float * input = static_cast<const float *>(input_buffer); + for (int i = 0; i < 32; ++i) { + // First part (16 frames) contains the reamaining input samples + // from previous iteration (since they were more). + ASSERT_FLOAT_EQ(input[i], 0.01 * (i + 32)); + } + for (int i = 0; i < 16; ++i) { + // next part (8 frames) contains the new input buffer. + ASSERT_FLOAT_EQ(input[i + 32], 0.01 * i); + // last part (8 frames) contains silence. + ASSERT_FLOAT_EQ(input[i + 32 + 16], 0.0); + } + }(); + } + return nframes; +} + +TEST(cubeb, passthrough_resampler_fill_input_left) +{ + const uint32_t channels = 2; + const uint32_t sample_rate = 44100; + int iteration = 0; + passthrough_resampler<float> resampler = passthrough_resampler<float>( + nullptr, passthrough_resampler_fill_input_left, &iteration, channels, + sample_rate); + + long input_frame_count = 48; // 32 + 16 + const long output_frame_count = 32; + float input[96] = {}; + float output[64] = {}; + for (uint32_t i = 0; i < input_frame_count * channels; ++i) { + input[i] = 0.01 * i; + } + + // 1st iteration, add the extra input. + iteration = 1; + long got = + resampler.fill(input, &input_frame_count, output, output_frame_count); + ASSERT_EQ(got, output_frame_count); + // Input frames used must be equal to output frames. + ASSERT_EQ(input_frame_count, output_frame_count); + + // 2st iteration, use the extra input from previous iteration, + // 16 frames are remaining in the input buffer. + input_frame_count = 32; // we need 16 input frames but we get more; + iteration = 2; + got = resampler.fill(input, &input_frame_count, output, output_frame_count); + ASSERT_EQ(got, output_frame_count); + // Input frames used must be equal to output frames. + ASSERT_EQ(input_frame_count, output_frame_count); + + // 3rd iteration, use the extra input from previous iteration. + // 16 frames are remaining in the input buffer. + input_frame_count = 16 - 8; // We need 16 more input frames but we only get 8. + iteration = 3; + got = resampler.fill(input, &input_frame_count, output, output_frame_count); + ASSERT_EQ(got, output_frame_count); + // Input frames used are less than the output frames due to glitch. + ASSERT_EQ(input_frame_count, output_frame_count - 8); +} + +TEST(cubeb, individual_methods) +{ + const uint32_t channels = 2; + const uint32_t sample_rate = 44100; + const uint32_t frames = 256; + + delay_line<float> dl(10, channels, sample_rate); + uint32_t frames_needed1 = dl.input_needed_for_output(0); + ASSERT_EQ(frames_needed1, 0u); + + cubeb_resampler_speex_one_way<float> one_way( + channels, sample_rate, sample_rate, CUBEB_RESAMPLER_QUALITY_DEFAULT); + float buffer[channels * frames] = {0.0}; + // Add all frames in the resampler's internal buffer. + one_way.input(buffer, frames); + // Ask for less than the existing frames, this would create a uint overlflow + // without the fix. + uint32_t frames_needed2 = one_way.input_needed_for_output(0); + ASSERT_EQ(frames_needed2, 0u); +} + +#undef NOMINMAX +#undef DUMP_ARRAYS |