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Diffstat (limited to 'dom/media/gtest/TestDynamicResampler.cpp')
| -rw-r--r-- | dom/media/gtest/TestDynamicResampler.cpp | 1556 |
1 files changed, 1556 insertions, 0 deletions
diff --git a/dom/media/gtest/TestDynamicResampler.cpp b/dom/media/gtest/TestDynamicResampler.cpp new file mode 100644 index 0000000000..4db0bd4854 --- /dev/null +++ b/dom/media/gtest/TestDynamicResampler.cpp @@ -0,0 +1,1556 @@ +/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/ +/* This Source Code Form is subject to the terms of the Mozilla Public + * License, v. 2.0. If a copy of the MPL was not distributed with this file, + * You can obtain one at http://mozilla.org/MPL/2.0/. */ + +#include "gmock/gmock.h" +#include "gtest/gtest-printers.h" +#include "gtest/gtest.h" + +#include "DynamicResampler.h" + +#include "nsContentUtils.h" + +using namespace mozilla; + +TEST(TestDynamicResampler, SameRates_Float1) +{ + const uint32_t in_frames = 100; + const uint32_t out_frames = 100; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + + // float in_ch1[] = {.1, .2, .3, .4, .5, .6, .7, .8, .9, 1.0}; + // float in_ch2[] = {.1, .2, .3, .4, .5, .6, .7, .8, .9, 1.0}; + float in_ch1[in_frames] = {}; + float in_ch2[in_frames] = {}; + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + float out_ch1[out_frames] = {}; + float out_ch2[out_frames] = {}; + + // Warm up with zeros + dr.AppendInput(in_buffer, in_frames); + uint32_t out_frames_used = out_frames; + bool rv = dr.Resample(out_ch1, &out_frames_used, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames_used, out_frames); + rv = dr.Resample(out_ch2, &out_frames_used, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames_used, out_frames); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_FLOAT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_FLOAT_EQ(in_ch2[i], out_ch2[i]); + } + + // Continue with non zero + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch1[i] = in_ch2[i] = 0.01f * i; + } + dr.AppendInput(in_buffer, in_frames); + out_frames_used = out_frames; + rv = dr.Resample(out_ch1, &out_frames_used, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames_used, out_frames); + rv = dr.Resample(out_ch2, &out_frames_used, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames_used, out_frames); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_FLOAT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_FLOAT_EQ(in_ch2[i], out_ch2[i]); + } + + // No more frames in the input buffer + rv = dr.Resample(out_ch1, &out_frames_used, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames_used, 0u); + out_frames_used = 2; + rv = dr.Resample(out_ch2, &out_frames_used, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames_used, 0u); +} + +TEST(TestDynamicResampler, SameRates_Short1) +{ + uint32_t in_frames = 2; + uint32_t out_frames = 2; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + + short in_ch1[] = {1, 2, 3}; + short in_ch2[] = {4, 5, 6}; + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + short out_ch1[3] = {}; + short out_ch2[3] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_EQ(in_ch2[i], out_ch2[i]); + } + + // No more frames in the input buffer + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + out_frames = 2; + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); +} + +TEST(TestDynamicResampler, SameRates_Float2) +{ + uint32_t in_frames = 3; + uint32_t out_frames = 2; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + float in_ch1[] = {0.1, 0.2, 0.3}; + float in_ch2[] = {0.4, 0.5, 0.6}; + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + float out_ch1[3] = {}; + float out_ch2[3] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_FLOAT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_FLOAT_EQ(in_ch2[i], out_ch2[i]); + } + + out_frames = 1; + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 1u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 1u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_FLOAT_EQ(in_ch1[i + 2], out_ch1[i]); + EXPECT_FLOAT_EQ(in_ch2[i + 2], out_ch2[i]); + } + + // No more frames, the input buffer has drained + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + out_frames = 1; + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); +} + +TEST(TestDynamicResampler, SameRates_Short2) +{ + uint32_t in_frames = 3; + uint32_t out_frames = 2; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + + short in_ch1[] = {1, 2, 3}; + short in_ch2[] = {4, 5, 6}; + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + short out_ch1[3] = {}; + short out_ch2[3] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 2u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_EQ(in_ch2[i], out_ch2[i]); + } + + out_frames = 1; + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 1u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 1u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_EQ(in_ch1[i + 2], out_ch1[i]); + EXPECT_EQ(in_ch2[i + 2], out_ch2[i]); + } + + // No more frames, the input buffer has drained + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + out_frames = 1; + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); +} + +TEST(TestDynamicResampler, SameRates_Float3) +{ + uint32_t in_frames = 2; + uint32_t out_frames = 3; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + float in_ch1[] = {0.1, 0.2, 0.3}; + float in_ch2[] = {0.4, 0.5, 0.6}; + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + float out_ch1[3] = {}; + float out_ch2[3] = {}; + + // Not enough frames in the input buffer + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + out_frames = 3; + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + + // Add one more frame + in_buffer[0] = in_ch1 + 2; + in_buffer[1] = in_ch2 + 2; + dr.AppendInput(in_buffer, 1); + out_frames = 3; + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 3u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 3u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_FLOAT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_FLOAT_EQ(in_ch2[i], out_ch2[i]); + } +} + +TEST(TestDynamicResampler, SameRates_Short3) +{ + uint32_t in_frames = 2; + uint32_t out_frames = 3; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + + short in_ch1[] = {1, 2, 3}; + short in_ch2[] = {4, 5, 6}; + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + short out_ch1[3] = {}; + short out_ch2[3] = {}; + + // Not enough frames in the input buffer + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + out_frames = 3; + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_FALSE(rv); + EXPECT_EQ(out_frames, 0u); + + // Add one more frame + in_buffer[0] = in_ch1 + 2; + in_buffer[1] = in_ch2 + 2; + dr.AppendInput(in_buffer, 1); + out_frames = 3; + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 3u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 3u); + for (uint32_t i = 0; i < out_frames; ++i) { + EXPECT_EQ(in_ch1[i], out_ch1[i]); + EXPECT_EQ(in_ch2[i], out_ch2[i]); + } +} + +TEST(TestDynamicResampler, UpdateOutRate_Float) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 20; + + DynamicResampler dr(in_rate, out_rate, pre_buffer); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + + float in_ch1[10] = {}; + float in_ch2[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch1[i] = in_ch2[i] = 0.01f * i; + } + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + float out_ch1[40] = {}; + float out_ch2[40] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 40u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 40u); + for (uint32_t i = 0; i < out_frames; ++i) { + // Only pre buffered data reach output + EXPECT_FLOAT_EQ(out_ch1[i], 0.0); + EXPECT_FLOAT_EQ(out_ch2[i], 0.0); + } + + // Update out rate + out_rate = 44100; + dr.UpdateResampler(out_rate, channels); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + out_frames = in_frames * out_rate / in_rate; + EXPECT_EQ(out_frames, 18u); + // Even if we provide no input if we have enough buffered input, we can create + // output + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 18u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 18u); +} + +TEST(TestDynamicResampler, UpdateOutRate_Short) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 20; + + DynamicResampler dr(in_rate, out_rate, pre_buffer); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + + short in_ch1[10] = {}; + short in_ch2[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch1[i] = in_ch2[i] = i; + } + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + short out_ch1[40] = {}; + short out_ch2[40] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 40u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 40u); + for (uint32_t i = 0; i < out_frames; ++i) { + // Only pre buffered data reach output + EXPECT_EQ(out_ch1[i], 0.0); + EXPECT_EQ(out_ch2[i], 0.0); + } + + // Update out rate + out_rate = 44100; + dr.UpdateResampler(out_rate, channels); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + out_frames = in_frames * out_rate / in_rate; + EXPECT_EQ(out_frames, 18u); + // Even if we provide no input if we have enough buffered input, we can create + // output + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 18u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 18u); +} + +TEST(TestDynamicResampler, BigRangeOutRates_Float) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 10; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + uint32_t pre_buffer = 20; + + DynamicResampler dr(in_rate, out_rate, pre_buffer); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + const uint32_t in_capacity = 40; + float in_ch1[in_capacity] = {}; + float in_ch2[in_capacity] = {}; + for (uint32_t i = 0; i < in_capacity; ++i) { + in_ch1[i] = in_ch2[i] = 0.01f * i; + } + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + const uint32_t out_capacity = 1000; + float out_ch1[out_capacity] = {}; + float out_ch2[out_capacity] = {}; + + for (uint32_t rate = 10000; rate < 90000; ++rate) { + out_rate = rate; + dr.UpdateResampler(out_rate, channels); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), channels); + in_frames = 20; // more than we need + out_frames = in_frames * out_rate / in_rate; + uint32_t expected_out_frames = out_frames; + for (uint32_t y = 0; y < 2; ++y) { + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, expected_out_frames); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, expected_out_frames); + } + } +} + +TEST(TestDynamicResampler, BigRangeOutRates_Short) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 10; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 44100; + uint32_t pre_buffer = 20; + + DynamicResampler dr(in_rate, out_rate, pre_buffer); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + + const uint32_t in_capacity = 40; + short in_ch1[in_capacity] = {}; + short in_ch2[in_capacity] = {}; + for (uint32_t i = 0; i < in_capacity; ++i) { + in_ch1[i] = in_ch2[i] = i; + } + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + const uint32_t out_capacity = 1000; + short out_ch1[out_capacity] = {}; + short out_ch2[out_capacity] = {}; + + for (uint32_t rate = 10000; rate < 90000; ++rate) { + out_rate = rate; + dr.UpdateResampler(out_rate, channels); + in_frames = 20; // more than we need + out_frames = in_frames * out_rate / in_rate; + uint32_t expected_out_frames = out_frames; + for (uint32_t y = 0; y < 2; ++y) { + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, expected_out_frames); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, expected_out_frames); + } + } +} + +TEST(TestDynamicResampler, UpdateChannels_Float) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 10; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 48000; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + float in_ch1[10] = {}; + float in_ch2[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch1[i] = in_ch2[i] = 0.01f * i; + } + AutoTArray<const float*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + float out_ch1[10] = {}; + float out_ch2[10] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + + // Add 3rd channel + dr.UpdateResampler(out_rate, 3); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), 3u); + + float in_ch3[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch3[i] = 0.01f * i; + } + in_buffer.AppendElement(); + in_buffer[2] = in_ch3; + float out_ch3[10] = {}; + + dr.AppendInput(in_buffer, in_frames); + + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch3, &out_frames, 2); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + + float in_ch4[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch3[i] = 0.01f * i; + } + in_buffer.AppendElement(); + in_buffer[3] = in_ch4; + float out_ch4[10] = {}; + + dr.UpdateResampler(out_rate, 4); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), 4u); + dr.AppendInput(in_buffer, in_frames); + + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch3, &out_frames, 2); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch4, &out_frames, 3); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); +} + +TEST(TestDynamicResampler, UpdateChannels_Short) +{ + uint32_t in_frames = 10; + uint32_t out_frames = 10; + uint32_t channels = 2; + uint32_t in_rate = 44100; + uint32_t out_rate = 48000; + + DynamicResampler dr(in_rate, out_rate); + dr.SetSampleFormat(AUDIO_FORMAT_S16); + + short in_ch1[10] = {}; + short in_ch2[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch1[i] = in_ch2[i] = i; + } + AutoTArray<const short*, 2> in_buffer; + in_buffer.AppendElements(channels); + in_buffer[0] = in_ch1; + in_buffer[1] = in_ch2; + + short out_ch1[10] = {}; + short out_ch2[10] = {}; + + dr.AppendInput(in_buffer, in_frames); + bool rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + + // Add 3rd channel + dr.UpdateResampler(out_rate, 3); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), 3u); + + short in_ch3[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch3[i] = i; + } + in_buffer.AppendElement(); + in_buffer[2] = in_ch3; + short out_ch3[10] = {}; + + dr.AppendInput(in_buffer, in_frames); + + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch3, &out_frames, 2); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + + // Check update with AudioSegment + short in_ch4[10] = {}; + for (uint32_t i = 0; i < in_frames; ++i) { + in_ch3[i] = i; + } + in_buffer.AppendElement(); + in_buffer[3] = in_ch4; + short out_ch4[10] = {}; + + dr.UpdateResampler(out_rate, 4); + EXPECT_EQ(dr.GetOutRate(), out_rate); + EXPECT_EQ(dr.GetChannels(), 4u); + dr.AppendInput(in_buffer, in_frames); + + rv = dr.Resample(out_ch1, &out_frames, 0); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch2, &out_frames, 1); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch3, &out_frames, 2); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); + rv = dr.Resample(out_ch4, &out_frames, 3); + EXPECT_TRUE(rv); + EXPECT_EQ(out_frames, 10u); +} + +TEST(TestAudioChunkList, Basic1) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + AudioChunkList list(256, 2, testPrincipal); + list.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + EXPECT_EQ(list.ChunkCapacity(), 128u); + EXPECT_EQ(list.TotalCapacity(), 256u); + + AudioChunk& c1 = list.GetNext(); + float* c1_ch1 = c1.ChannelDataForWrite<float>(0); + float* c1_ch2 = c1.ChannelDataForWrite<float>(1); + EXPECT_EQ(c1.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c1.mBufferFormat, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + c1_ch1[i] = c1_ch2[i] = 0.01f * static_cast<float>(i); + } + AudioChunk& c2 = list.GetNext(); + EXPECT_EQ(c2.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c2.mBufferFormat, AUDIO_FORMAT_FLOAT32); + EXPECT_NE(c1.mBuffer.get(), c2.mBuffer.get()); + AudioChunk& c3 = list.GetNext(); + EXPECT_EQ(c3.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c3.mBufferFormat, AUDIO_FORMAT_FLOAT32); + // Cycle + EXPECT_EQ(c1.mBuffer.get(), c3.mBuffer.get()); + float* c3_ch1 = c3.ChannelDataForWrite<float>(0); + float* c3_ch2 = c3.ChannelDataForWrite<float>(1); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + EXPECT_FLOAT_EQ(c1_ch1[i], c3_ch1[i]); + EXPECT_FLOAT_EQ(c1_ch2[i], c3_ch2[i]); + } +} + +TEST(TestAudioChunkList, Basic2) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + AudioChunkList list(256, 2, testPrincipal); + list.SetSampleFormat(AUDIO_FORMAT_S16); + EXPECT_EQ(list.ChunkCapacity(), 256u); + EXPECT_EQ(list.TotalCapacity(), 512u); + + AudioChunk& c1 = list.GetNext(); + EXPECT_EQ(c1.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c1.mBufferFormat, AUDIO_FORMAT_S16); + short* c1_ch1 = c1.ChannelDataForWrite<short>(0); + short* c1_ch2 = c1.ChannelDataForWrite<short>(1); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + c1_ch1[i] = c1_ch2[i] = static_cast<short>(i); + } + AudioChunk& c2 = list.GetNext(); + EXPECT_EQ(c2.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c2.mBufferFormat, AUDIO_FORMAT_S16); + EXPECT_NE(c1.mBuffer.get(), c2.mBuffer.get()); + AudioChunk& c3 = list.GetNext(); + EXPECT_EQ(c3.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c3.mBufferFormat, AUDIO_FORMAT_S16); + AudioChunk& c4 = list.GetNext(); + EXPECT_EQ(c4.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c4.mBufferFormat, AUDIO_FORMAT_S16); + // Cycle + AudioChunk& c5 = list.GetNext(); + EXPECT_EQ(c5.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c5.mBufferFormat, AUDIO_FORMAT_S16); + EXPECT_EQ(c1.mBuffer.get(), c5.mBuffer.get()); + short* c5_ch1 = c5.ChannelDataForWrite<short>(0); + short* c5_ch2 = c5.ChannelDataForWrite<short>(1); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + EXPECT_EQ(c1_ch1[i], c5_ch1[i]); + EXPECT_EQ(c1_ch2[i], c5_ch2[i]); + } +} + +TEST(TestAudioChunkList, Basic3) +{ + AudioChunkList list(260, 2, PRINCIPAL_HANDLE_NONE); + list.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + EXPECT_EQ(list.ChunkCapacity(), 128u); + EXPECT_EQ(list.TotalCapacity(), 256u + 128u); + + AudioChunk& c1 = list.GetNext(); + AudioChunk& c2 = list.GetNext(); + EXPECT_NE(c1.mBuffer.get(), c2.mBuffer.get()); + AudioChunk& c3 = list.GetNext(); + EXPECT_NE(c1.mBuffer.get(), c3.mBuffer.get()); + AudioChunk& c4 = list.GetNext(); + EXPECT_EQ(c1.mBuffer.get(), c4.mBuffer.get()); +} + +TEST(TestAudioChunkList, Basic4) +{ + AudioChunkList list(260, 2, PRINCIPAL_HANDLE_NONE); + list.SetSampleFormat(AUDIO_FORMAT_S16); + EXPECT_EQ(list.ChunkCapacity(), 256u); + EXPECT_EQ(list.TotalCapacity(), 512u + 256u); + + AudioChunk& c1 = list.GetNext(); + AudioChunk& c2 = list.GetNext(); + EXPECT_NE(c1.mBuffer.get(), c2.mBuffer.get()); + AudioChunk& c3 = list.GetNext(); + EXPECT_NE(c1.mBuffer.get(), c3.mBuffer.get()); + AudioChunk& c4 = list.GetNext(); + EXPECT_EQ(c1.mBuffer.get(), c4.mBuffer.get()); +} + +TEST(TestAudioChunkList, UpdateChannels) +{ + AudioChunkList list(256, 2, PRINCIPAL_HANDLE_NONE); + list.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + AudioChunk& c1 = list.GetNext(); + AudioChunk& c2 = list.GetNext(); + EXPECT_EQ(c1.ChannelCount(), 2u); + EXPECT_EQ(c2.ChannelCount(), 2u); + + // Update to Quad + list.Update(4); + + AudioChunk& c3 = list.GetNext(); + AudioChunk& c4 = list.GetNext(); + EXPECT_EQ(c3.ChannelCount(), 4u); + EXPECT_EQ(c4.ChannelCount(), 4u); +} + +TEST(TestAudioChunkList, UpdateBetweenMonoAndStereo) +{ + AudioChunkList list(256, 2, PRINCIPAL_HANDLE_NONE); + list.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + AudioChunk& c1 = list.GetNext(); + float* c1_ch1 = c1.ChannelDataForWrite<float>(0); + float* c1_ch2 = c1.ChannelDataForWrite<float>(1); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + c1_ch1[i] = c1_ch2[i] = 0.01f * static_cast<float>(i); + } + + AudioChunk& c2 = list.GetNext(); + EXPECT_EQ(c1.ChannelCount(), 2u); + EXPECT_EQ(c2.ChannelCount(), 2u); + + // Downmix to mono + list.Update(1); + + AudioChunk& c3 = list.GetNext(); + float* c3_ch1 = c3.ChannelDataForWrite<float>(0); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + EXPECT_FLOAT_EQ(c3_ch1[i], c1_ch1[i]); + } + + AudioChunk& c4 = list.GetNext(); + EXPECT_EQ(c3.ChannelCount(), 1u); + EXPECT_EQ(c4.ChannelCount(), 1u); + EXPECT_EQ(static_cast<SharedChannelArrayBuffer<float>*>(c3.mBuffer.get()) + ->mBuffers[0] + .Length(), + list.ChunkCapacity()); + + // Upmix to stereo + list.Update(2); + + AudioChunk& c5 = list.GetNext(); + AudioChunk& c6 = list.GetNext(); + EXPECT_EQ(c5.ChannelCount(), 2u); + EXPECT_EQ(c6.ChannelCount(), 2u); + EXPECT_EQ(static_cast<SharedChannelArrayBuffer<float>*>(c5.mBuffer.get()) + ->mBuffers[0] + .Length(), + list.ChunkCapacity()); + EXPECT_EQ(static_cast<SharedChannelArrayBuffer<float>*>(c5.mBuffer.get()) + ->mBuffers[1] + .Length(), + list.ChunkCapacity()); + + // Downmix to mono + list.Update(1); + + AudioChunk& c7 = list.GetNext(); + float* c7_ch1 = c7.ChannelDataForWrite<float>(0); + for (uint32_t i = 0; i < list.ChunkCapacity(); ++i) { + EXPECT_FLOAT_EQ(c7_ch1[i], c1_ch1[i]); + } + + AudioChunk& c8 = list.GetNext(); + EXPECT_EQ(c7.ChannelCount(), 1u); + EXPECT_EQ(c8.ChannelCount(), 1u); + EXPECT_EQ(static_cast<SharedChannelArrayBuffer<float>*>(c7.mBuffer.get()) + ->mBuffers[0] + .Length(), + list.ChunkCapacity()); +} + +TEST(TestAudioChunkList, ConsumeAndForget) +{ + AudioSegment s; + AudioChunkList list(256, 2, PRINCIPAL_HANDLE_NONE); + list.SetSampleFormat(AUDIO_FORMAT_FLOAT32); + + AudioChunk& c1 = list.GetNext(); + AudioChunk tmp1 = c1; + s.AppendAndConsumeChunk(std::move(tmp1)); + EXPECT_FALSE(c1.mBuffer.get() == nullptr); + EXPECT_EQ(c1.ChannelData<float>().Length(), 2u); + + AudioChunk& c2 = list.GetNext(); + AudioChunk tmp2 = c2; + s.AppendAndConsumeChunk(std::move(tmp2)); + EXPECT_FALSE(c2.mBuffer.get() == nullptr); + EXPECT_EQ(c2.ChannelData<float>().Length(), 2u); + + s.ForgetUpTo(256); + list.GetNext(); + list.GetNext(); +} + +template <class T> +AudioChunk CreateAudioChunk(uint32_t aFrames, uint32_t aChannels, + AudioSampleFormat aSampleFormat) { + AudioChunk chunk; + nsTArray<nsTArray<T>> buffer; + buffer.AppendElements(aChannels); + + nsTArray<const T*> bufferPtrs; + bufferPtrs.AppendElements(aChannels); + + for (uint32_t i = 0; i < aChannels; ++i) { + T* ptr = buffer[i].AppendElements(aFrames); + bufferPtrs[i] = ptr; + for (uint32_t j = 0; j < aFrames; ++j) { + if (aSampleFormat == AUDIO_FORMAT_FLOAT32) { + ptr[j] = 0.01 * j; + } else { + ptr[j] = j; + } + } + } + + chunk.mBuffer = new mozilla::SharedChannelArrayBuffer(std::move(buffer)); + chunk.mBufferFormat = aSampleFormat; + chunk.mChannelData.AppendElements(aChannels); + for (uint32_t i = 0; i < aChannels; ++i) { + chunk.mChannelData[i] = bufferPtrs[i]; + } + chunk.mDuration = aFrames; + return chunk; +} + +template <class T> +AudioSegment CreateAudioSegment(uint32_t aFrames, uint32_t aChannels, + AudioSampleFormat aSampleFormat) { + AudioSegment segment; + AudioChunk chunk = CreateAudioChunk<T>(aFrames, aChannels, aSampleFormat); + segment.AppendAndConsumeChunk(std::move(chunk)); + return segment; +} + +TEST(TestAudioResampler, OutAudioSegment_Float) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 21; + + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioSegment inSegment = + CreateAudioSegment<float>(in_frames, channels, AUDIO_FORMAT_FLOAT32); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 40); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s.IsNull()); + EXPECT_TRUE(!s.IsEmpty()); + + for (AudioSegment::ChunkIterator ci(s); !ci.IsEnded(); ci.Next()) { + AudioChunk& c = *ci; + EXPECT_EQ(c.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c.ChannelCount(), 2u); + for (uint32_t i = 0; i < out_frames; ++i) { + // Only pre buffered data reach output + EXPECT_FLOAT_EQ(c.ChannelData<float>()[0][i], 0.0); + EXPECT_FLOAT_EQ(c.ChannelData<float>()[1][i], 0.0); + } + } + + // Update out rate + out_rate = 44100; + dr.UpdateOutRate(out_rate); + out_frames = in_frames * out_rate / in_rate; + EXPECT_EQ(out_frames, 18u); + // Even if we provide no input if we have enough buffered input, we can create + // output + AudioSegment s1 = dr.Resample(out_frames); + EXPECT_EQ(s1.GetDuration(), out_frames); + EXPECT_EQ(s1.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s1.IsNull()); + EXPECT_TRUE(!s1.IsEmpty()); + for (AudioSegment::ConstChunkIterator ci(s1); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, OutAudioSegment_Short) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 21; + + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioSegment inSegment = + CreateAudioSegment<short>(in_frames, channels, AUDIO_FORMAT_S16); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 40); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s.IsNull()); + EXPECT_TRUE(!s.IsEmpty()); + + for (AudioSegment::ChunkIterator ci(s); !ci.IsEnded(); ci.Next()) { + AudioChunk& c = *ci; + EXPECT_EQ(c.mPrincipalHandle, testPrincipal); + EXPECT_EQ(c.ChannelCount(), 2u); + for (uint32_t i = 0; i < out_frames; ++i) { + // Only pre buffered data reach output + EXPECT_FLOAT_EQ(c.ChannelData<short>()[0][i], 0.0); + EXPECT_FLOAT_EQ(c.ChannelData<short>()[1][i], 0.0); + } + } + + // Update out rate + out_rate = 44100; + dr.UpdateOutRate(out_rate); + out_frames = in_frames * out_rate / in_rate; + EXPECT_EQ(out_frames, 18u); + // Even if we provide no input if we have enough buffered input, we can create + // output + AudioSegment s1 = dr.Resample(out_frames); + EXPECT_EQ(s1.GetDuration(), out_frames); + EXPECT_EQ(s1.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s1.IsNull()); + EXPECT_TRUE(!s1.IsEmpty()); + for (AudioSegment::ConstChunkIterator ci(s1); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, OutAudioSegmentFail_Float) +{ + const uint32_t in_frames = 130; + const uint32_t out_frames = 300; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 5; + + AudioResampler dr(in_rate, out_rate, pre_buffer, PRINCIPAL_HANDLE_NONE); + AudioSegment inSegment = + CreateAudioSegment<float>(in_frames, channels, AUDIO_FORMAT_FLOAT32); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 0); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(s.IsNull()); + EXPECT_TRUE(s.IsEmpty()); +} + +TEST(TestAudioResampler, InAudioSegment_Float) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 10; + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioSegment inSegment; + + AudioChunk chunk1; + chunk1.SetNull(in_frames / 2); + inSegment.AppendAndConsumeChunk(std::move(chunk1)); + + AudioChunk chunk2; + nsTArray<nsTArray<float>> buffer; + buffer.AppendElements(channels); + + nsTArray<const float*> bufferPtrs; + bufferPtrs.AppendElements(channels); + + for (uint32_t i = 0; i < channels; ++i) { + float* ptr = buffer[i].AppendElements(5); + bufferPtrs[i] = ptr; + for (uint32_t j = 0; j < 5; ++j) { + ptr[j] = 0.01f * j; + } + } + + chunk2.mBuffer = new mozilla::SharedChannelArrayBuffer(std::move(buffer)); + chunk2.mBufferFormat = AUDIO_FORMAT_FLOAT32; + chunk2.mChannelData.AppendElements(channels); + for (uint32_t i = 0; i < channels; ++i) { + chunk2.mChannelData[i] = bufferPtrs[i]; + } + chunk2.mDuration = in_frames / 2; + inSegment.AppendAndConsumeChunk(std::move(chunk2)); + + dr.AppendInput(inSegment); + AudioSegment outSegment = dr.Resample(out_frames); + // Faild because the first chunk is ignored + EXPECT_EQ(outSegment.GetDuration(), 0u); + EXPECT_EQ(outSegment.MaxChannelCount(), 0u); + + // Add the 5 more frames that are missing + dr.AppendInput(inSegment); + AudioSegment outSegment2 = dr.Resample(out_frames); + EXPECT_EQ(outSegment2.GetDuration(), 40u); + EXPECT_EQ(outSegment2.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(outSegment2); !ci.IsEnded(); + ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, InAudioSegment_Short) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 10; + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioSegment inSegment; + + // The null chunk at the beginning will be ignored. + AudioChunk chunk1; + chunk1.SetNull(in_frames / 2); + inSegment.AppendAndConsumeChunk(std::move(chunk1)); + + AudioChunk chunk2; + nsTArray<nsTArray<short>> buffer; + buffer.AppendElements(channels); + + nsTArray<const short*> bufferPtrs; + bufferPtrs.AppendElements(channels); + + for (uint32_t i = 0; i < channels; ++i) { + short* ptr = buffer[i].AppendElements(5); + bufferPtrs[i] = ptr; + for (uint32_t j = 0; j < 5; ++j) { + ptr[j] = j; + } + } + + chunk2.mBuffer = new mozilla::SharedChannelArrayBuffer(std::move(buffer)); + chunk2.mBufferFormat = AUDIO_FORMAT_S16; + chunk2.mChannelData.AppendElements(channels); + for (uint32_t i = 0; i < channels; ++i) { + chunk2.mChannelData[i] = bufferPtrs[i]; + } + chunk2.mDuration = in_frames / 2; + inSegment.AppendAndConsumeChunk(std::move(chunk2)); + + dr.AppendInput(inSegment); + AudioSegment outSegment = dr.Resample(out_frames); + // Faild because the first chunk is ignored + EXPECT_EQ(outSegment.GetDuration(), 0u); + EXPECT_EQ(outSegment.MaxChannelCount(), 0u); + + dr.AppendInput(inSegment); + AudioSegment outSegment2 = dr.Resample(out_frames); + EXPECT_EQ(outSegment2.GetDuration(), 40u); + EXPECT_EQ(outSegment2.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(outSegment2); !ci.IsEnded(); + ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_MonoToStereo) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + // uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 0; + + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioChunk monoChunk = + CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32); + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(monoChunk)); + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 40); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s.IsNull()); + EXPECT_TRUE(!s.IsEmpty()); + EXPECT_EQ(s.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(s); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_StereoToMono) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + // uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 0; + + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioChunk monoChunk = + CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32); + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + inSegment.AppendAndConsumeChunk(std::move(monoChunk)); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 40); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s.IsNull()); + EXPECT_TRUE(!s.IsEmpty()); + EXPECT_EQ(s.MaxChannelCount(), 1u); + for (AudioSegment::ConstChunkIterator ci(s); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_StereoToQuad) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + // uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + uint32_t pre_buffer = 0; + + AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal); + + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + AudioChunk quadChunk = + CreateAudioChunk<float>(in_frames, 4, AUDIO_FORMAT_FLOAT32); + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + inSegment.AppendAndConsumeChunk(std::move(quadChunk)); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 0); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(s.IsNull()); + EXPECT_TRUE(s.IsEmpty()); + + AudioSegment s2 = dr.Resample(out_frames / 2); + EXPECT_EQ(s2.GetDuration(), out_frames / 2); + EXPECT_EQ(s2.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s2.IsNull()); + EXPECT_TRUE(!s2.IsEmpty()); + for (AudioSegment::ConstChunkIterator ci(s2); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_QuadToStereo) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_frames = 10; + uint32_t out_frames = 40; + // uint32_t channels = 2; + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + AudioResampler dr(in_rate, out_rate, 0, testPrincipal); + + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + AudioChunk quadChunk = + CreateAudioChunk<float>(in_frames, 4, AUDIO_FORMAT_FLOAT32); + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(quadChunk)); + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + dr.AppendInput(inSegment); + + AudioSegment s = dr.Resample(out_frames); + EXPECT_EQ(s.GetDuration(), 0); + EXPECT_EQ(s.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(s.IsNull()); + EXPECT_TRUE(s.IsEmpty()); + + AudioSegment s2 = dr.Resample(out_frames / 2); + EXPECT_EQ(s2.GetDuration(), out_frames / 2); + EXPECT_EQ(s2.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s2.IsNull()); + EXPECT_TRUE(!s2.IsEmpty()); + for (AudioSegment::ConstChunkIterator ci(s2); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +void printAudioSegment(const AudioSegment& segment); + +TEST(TestAudioResampler, ChannelChange_Discontinuity) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + const float amplitude = 0.5; + const float frequency = 200; + const float phase = 0.0; + float time = 0.0; + const float deltaTime = 1.0f / static_cast<float>(in_rate); + + uint32_t in_frames = in_rate / 100; + uint32_t out_frames = out_rate / 100; + AudioResampler dr(in_rate, out_rate, 0, testPrincipal); + + AudioChunk monoChunk = + CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < monoChunk.GetDuration(); ++i) { + double value = amplitude * sin(2 * M_PI * frequency * time + phase); + monoChunk.ChannelDataForWrite<float>(0)[i] = static_cast<float>(value); + time += deltaTime; + } + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < stereoChunk.GetDuration(); ++i) { + double value = amplitude * sin(2 * M_PI * frequency * time + phase); + stereoChunk.ChannelDataForWrite<float>(0)[i] = static_cast<float>(value); + if (stereoChunk.ChannelCount() == 2) { + stereoChunk.ChannelDataForWrite<float>(1)[i] = value; + } + time += deltaTime; + } + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + // printAudioSegment(inSegment); + + dr.AppendInput(inSegment); + AudioSegment s = dr.Resample(out_frames); + // printAudioSegment(s); + + AudioSegment inSegment2; + inSegment2.AppendAndConsumeChunk(std::move(monoChunk)); + // The resampler here is updated due to the channel change and that creates + // discontinuity. + dr.AppendInput(inSegment2); + AudioSegment s2 = dr.Resample(out_frames); + // printAudioSegment(s2); + + EXPECT_EQ(s2.GetDuration(), 480); + EXPECT_EQ(s2.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s2.IsNull()); + EXPECT_TRUE(!s2.IsEmpty()); + EXPECT_EQ(s2.MaxChannelCount(), 1u); + for (AudioSegment::ConstChunkIterator ci(s2); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_Discontinuity2) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_rate = 24000; + uint32_t out_rate = 48000; + + const float amplitude = 0.5; + const float frequency = 200; + const float phase = 0.0; + float time = 0.0; + const float deltaTime = 1.0f / static_cast<float>(in_rate); + + uint32_t in_frames = in_rate / 100; + uint32_t out_frames = out_rate / 100; + AudioResampler dr(in_rate, out_rate, 10, testPrincipal); + + AudioChunk monoChunk = + CreateAudioChunk<float>(in_frames / 2, 1, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < monoChunk.GetDuration(); ++i) { + double value = amplitude * sin(2 * M_PI * frequency * time + phase); + monoChunk.ChannelDataForWrite<float>(0)[i] = static_cast<float>(value); + time += deltaTime; + } + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames / 2, 2, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < stereoChunk.GetDuration(); ++i) { + double value = amplitude * sin(2 * M_PI * frequency * time + phase); + stereoChunk.ChannelDataForWrite<float>(0)[i] = static_cast<float>(value); + if (stereoChunk.ChannelCount() == 2) { + stereoChunk.ChannelDataForWrite<float>(1)[i] = value; + } + time += deltaTime; + } + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(monoChunk)); + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + // printAudioSegment(inSegment); + + dr.AppendInput(inSegment); + AudioSegment s1 = dr.Resample(out_frames); + // printAudioSegment(s1); + + EXPECT_EQ(s1.GetDuration(), 480); + EXPECT_EQ(s1.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s1.IsNull()); + EXPECT_TRUE(!s1.IsEmpty()); + EXPECT_EQ(s1.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(s1); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } + + // The resampler here is updated due to the channel change and that creates + // discontinuity. + dr.AppendInput(inSegment); + AudioSegment s2 = dr.Resample(out_frames); + // printAudioSegment(s2); + + EXPECT_EQ(s2.GetDuration(), 480); + EXPECT_EQ(s2.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s2.IsNull()); + EXPECT_TRUE(!s2.IsEmpty()); + EXPECT_EQ(s2.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(s2); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} + +TEST(TestAudioResampler, ChannelChange_Discontinuity3) +{ + const PrincipalHandle testPrincipal = + MakePrincipalHandle(nsContentUtils::GetSystemPrincipal()); + + uint32_t in_rate = 48000; + uint32_t out_rate = 48000; + + const float amplitude = 0.5; + const float frequency = 200; + const float phase = 0.0; + float time = 0.0; + const float deltaTime = 1.0f / static_cast<float>(in_rate); + + uint32_t in_frames = in_rate / 100; + uint32_t out_frames = out_rate / 100; + AudioResampler dr(in_rate, out_rate, 10, testPrincipal); + + AudioChunk stereoChunk = + CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32); + for (uint32_t i = 0; i < stereoChunk.GetDuration(); ++i) { + double value = amplitude * sin(2 * M_PI * frequency * time + phase); + stereoChunk.ChannelDataForWrite<float>(0)[i] = static_cast<float>(value); + if (stereoChunk.ChannelCount() == 2) { + stereoChunk.ChannelDataForWrite<float>(1)[i] = value; + } + time += deltaTime; + } + + AudioSegment inSegment; + inSegment.AppendAndConsumeChunk(std::move(stereoChunk)); + // printAudioSegment(inSegment); + + dr.AppendInput(inSegment); + AudioSegment s = dr.Resample(out_frames); + // printAudioSegment(s); + + // The resampler here is updated due to the rate change. This is because the + // in and out rate was the same so a pass through logice was used. By updating + // the out rate to something different than the in rate, the resampler will + // start being use dand discontinuity will exist. + dr.UpdateOutRate(out_rate + 100); + dr.AppendInput(inSegment); + AudioSegment s2 = dr.Resample(out_frames); + // printAudioSegment(s2); + + EXPECT_EQ(s2.GetDuration(), 480); + EXPECT_EQ(s2.GetType(), MediaSegment::AUDIO); + EXPECT_TRUE(!s2.IsNull()); + EXPECT_TRUE(!s2.IsEmpty()); + EXPECT_EQ(s2.MaxChannelCount(), 2u); + for (AudioSegment::ConstChunkIterator ci(s2); !ci.IsEnded(); ci.Next()) { + EXPECT_EQ(ci->mPrincipalHandle, testPrincipal); + } +} |