blob: 9079bcd1ddd3a58d6b739d996e9b1250558fbf14 [file] [log] [blame]
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
namespace webrtc {
class SplitBySamplesTest : public ::testing::TestWithParam<NetEqDecoder> {
protected:
virtual void SetUp() {
decoder_type_ = GetParam();
switch (decoder_type_) {
case NetEqDecoder::kDecoderPCMu:
case NetEqDecoder::kDecoderPCMa:
bytes_per_ms_ = 8;
samples_per_ms_ = 8;
break;
case NetEqDecoder::kDecoderPCMu_2ch:
case NetEqDecoder::kDecoderPCMa_2ch:
bytes_per_ms_ = 2 * 8;
samples_per_ms_ = 8;
break;
case NetEqDecoder::kDecoderG722:
bytes_per_ms_ = 8;
samples_per_ms_ = 16;
break;
case NetEqDecoder::kDecoderPCM16B:
bytes_per_ms_ = 16;
samples_per_ms_ = 8;
break;
case NetEqDecoder::kDecoderPCM16Bwb:
bytes_per_ms_ = 32;
samples_per_ms_ = 16;
break;
case NetEqDecoder::kDecoderPCM16Bswb32kHz:
bytes_per_ms_ = 64;
samples_per_ms_ = 32;
break;
case NetEqDecoder::kDecoderPCM16Bswb48kHz:
bytes_per_ms_ = 96;
samples_per_ms_ = 48;
break;
case NetEqDecoder::kDecoderPCM16B_2ch:
bytes_per_ms_ = 2 * 16;
samples_per_ms_ = 8;
break;
case NetEqDecoder::kDecoderPCM16Bwb_2ch:
bytes_per_ms_ = 2 * 32;
samples_per_ms_ = 16;
break;
case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
bytes_per_ms_ = 2 * 64;
samples_per_ms_ = 32;
break;
case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
bytes_per_ms_ = 2 * 96;
samples_per_ms_ = 48;
break;
case NetEqDecoder::kDecoderPCM16B_5ch:
bytes_per_ms_ = 5 * 16;
samples_per_ms_ = 8;
break;
default:
assert(false);
break;
}
}
size_t bytes_per_ms_;
int samples_per_ms_;
NetEqDecoder decoder_type_;
};
// Test splitting sample-based payloads.
TEST_P(SplitBySamplesTest, PayloadSizes) {
constexpr uint32_t kBaseTimestamp = 0x12345678;
struct ExpectedSplit {
size_t payload_size_ms;
size_t num_frames;
// For simplicity. We only expect up to two packets per split.
size_t frame_sizes[2];
};
// The payloads are expected to be split as follows:
// 10 ms -> 10 ms
// 20 ms -> 20 ms
// 30 ms -> 30 ms
// 40 ms -> 20 + 20 ms
// 50 ms -> 25 + 25 ms
// 60 ms -> 30 + 30 ms
ExpectedSplit expected_splits[] = {{10, 1, {10}}, {20, 1, {20}},
{30, 1, {30}}, {40, 2, {20, 20}},
{50, 2, {25, 25}}, {60, 2, {30, 30}}};
for (const auto& expected_split : expected_splits) {
// The payload values are set to steadily increase (modulo 256), so that the
// resulting frames can be checked and we can be reasonably certain no
// sample was missed or repeated.
const auto generate_payload = [](size_t num_bytes) {
rtc::Buffer payload(num_bytes);
uint8_t value = 0;
// Allow wrap-around of value in counter below.
for (size_t i = 0; i != payload.size(); ++i, ++value) {
payload[i] = value;
}
return payload;
};
const auto results = LegacyEncodedAudioFrame::SplitBySamples(
nullptr,
generate_payload(expected_split.payload_size_ms * bytes_per_ms_),
kBaseTimestamp, bytes_per_ms_, samples_per_ms_);
EXPECT_EQ(expected_split.num_frames, results.size());
uint32_t expected_timestamp = kBaseTimestamp;
uint32_t expected_byte_offset = 0;
uint8_t value = 0;
for (size_t i = 0; i != expected_split.num_frames; ++i) {
const auto& result = results[i];
const LegacyEncodedAudioFrame* frame =
static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
const size_t length_bytes = expected_split.frame_sizes[i] * bytes_per_ms_;
EXPECT_EQ(length_bytes, frame->payload().size());
EXPECT_EQ(expected_timestamp, result.timestamp);
const rtc::Buffer& payload = frame->payload();
// Allow wrap-around of value in counter below.
for (size_t i = 0; i != payload.size(); ++i, ++value) {
ASSERT_EQ(value, payload[i]);
}
expected_timestamp += rtc::checked_cast<uint32_t>(
expected_split.frame_sizes[i] * samples_per_ms_);
expected_byte_offset += rtc::checked_cast<uint32_t>(length_bytes);
}
}
}
INSTANTIATE_TEST_CASE_P(
LegacyEncodedAudioFrame,
SplitBySamplesTest,
::testing::Values(NetEqDecoder::kDecoderPCMu,
NetEqDecoder::kDecoderPCMa,
NetEqDecoder::kDecoderPCMu_2ch,
NetEqDecoder::kDecoderPCMa_2ch,
NetEqDecoder::kDecoderG722,
NetEqDecoder::kDecoderPCM16B,
NetEqDecoder::kDecoderPCM16Bwb,
NetEqDecoder::kDecoderPCM16Bswb32kHz,
NetEqDecoder::kDecoderPCM16Bswb48kHz,
NetEqDecoder::kDecoderPCM16B_2ch,
NetEqDecoder::kDecoderPCM16Bwb_2ch,
NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch,
NetEqDecoder::kDecoderPCM16B_5ch));
} // namespace webrtc