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2ab97f6f8e
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
102 lines
3.4 KiB
C++
102 lines
3.4 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "absl/flags/flag.h"
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#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
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#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
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ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
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using ::testing::InitGoogleTest;
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namespace webrtc {
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namespace test {
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namespace {
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static const int kIsacBlockDurationMs = 30;
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static const int kIsacInputSamplingKhz = 16;
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static const int kIsacOutputSamplingKhz = 16;
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} // namespace
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class NetEqIsacQualityTest : public NetEqQualityTest {
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protected:
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NetEqIsacQualityTest();
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void SetUp() override;
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void TearDown() override;
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int EncodeBlock(int16_t* in_data,
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size_t block_size_samples,
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rtc::Buffer* payload,
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size_t max_bytes) override;
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private:
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ISACFIX_MainStruct* isac_encoder_;
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int bit_rate_kbps_;
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};
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NetEqIsacQualityTest::NetEqIsacQualityTest()
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: NetEqQualityTest(kIsacBlockDurationMs,
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kIsacInputSamplingKhz,
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kIsacOutputSamplingKhz,
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SdpAudioFormat("isac", 16000, 1)),
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isac_encoder_(NULL),
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bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)) {
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// Flag validation
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RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 10 &&
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absl::GetFlag(FLAGS_bit_rate_kbps) <= 32)
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<< "Invalid bit rate, should be between 10 and 32 kbps.";
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}
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void NetEqIsacQualityTest::SetUp() {
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ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
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// Create encoder memory.
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WebRtcIsacfix_Create(&isac_encoder_);
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ASSERT_TRUE(isac_encoder_ != NULL);
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EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(isac_encoder_, 1));
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// Set bitrate and block length.
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EXPECT_EQ(0, WebRtcIsacfix_Control(isac_encoder_, bit_rate_kbps_ * 1000,
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kIsacBlockDurationMs));
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NetEqQualityTest::SetUp();
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}
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void NetEqIsacQualityTest::TearDown() {
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// Free memory.
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EXPECT_EQ(0, WebRtcIsacfix_Free(isac_encoder_));
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NetEqQualityTest::TearDown();
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}
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int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
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size_t block_size_samples,
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rtc::Buffer* payload,
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size_t max_bytes) {
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// ISAC takes 10 ms for every call.
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const int subblocks = kIsacBlockDurationMs / 10;
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const int subblock_length = 10 * kIsacInputSamplingKhz;
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int value = 0;
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int pointer = 0;
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for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
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// The Isac encoder does not perform encoding (and returns 0) until it
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// receives a sequence of sub-blocks that amount to the frame duration.
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EXPECT_EQ(0, value);
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payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
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value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
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payload.data());
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return (value >= 0) ? static_cast<size_t>(value) : 0;
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});
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}
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EXPECT_GT(value, 0);
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return value;
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}
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TEST_F(NetEqIsacQualityTest, Test) {
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Simulate();
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}
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} // namespace test
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} // namespace webrtc
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