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// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "base/memory/ref_counted.h"
#include "media/cast/audio_receiver/audio_decoder.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace media {
namespace cast {
class AudioDecoderTest : public ::testing::Test {
protected:
AudioDecoderTest() {}
virtual ~AudioDecoderTest() {}
void Configure(const AudioReceiverConfig& audio_config) {
audio_decoder_.reset(new AudioDecoder(audio_config));
}
scoped_ptr<AudioDecoder> audio_decoder_;
};
TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
AudioReceiverConfig audio_config;
audio_config.rtp_payload_type = 127;
audio_config.frequency = 16000;
audio_config.channels = 1;
audio_config.codec = kPcm16;
audio_config.use_external_decoder = false;
Configure(audio_config);
RtpCastHeader rtp_header;
rtp_header.webrtc.header.payloadType = 127;
rtp_header.webrtc.header.sequenceNumber = 1234;
rtp_header.webrtc.header.timestamp = 0x87654321;
rtp_header.webrtc.header.ssrc = 0x12345678;
rtp_header.webrtc.header.paddingLength = 0;
rtp_header.webrtc.header.headerLength = 12;
rtp_header.webrtc.type.Audio.channel = 1;
rtp_header.webrtc.type.Audio.isCNG = false;
std::vector<int16> payload(640, 0x1234);
int number_of_10ms_blocks = 4;
int desired_frequency = 16000;
PcmAudioFrame audio_frame;
uint32 rtp_timestamp;
EXPECT_FALSE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
desired_frequency,
&audio_frame,
&rtp_timestamp));
uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
size_t payload_size = payload.size() * sizeof(int16);
audio_decoder_->IncomingParsedRtpPacket(payload_data,
payload_size, rtp_header);
EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
desired_frequency,
&audio_frame,
&rtp_timestamp));
EXPECT_EQ(1, audio_frame.channels);
EXPECT_EQ(16000, audio_frame.frequency);
EXPECT_EQ(640ul, audio_frame.samples.size());
// First 10 samples per channel are 0 from NetEq.
for (size_t i = 10; i < audio_frame.samples.size(); ++i) {
EXPECT_EQ(0x3412, audio_frame.samples[i]);
}
}
TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
AudioReceiverConfig audio_config;
audio_config.rtp_payload_type = 127;
audio_config.frequency = 16000;
audio_config.channels = 2;
audio_config.codec = kPcm16;
audio_config.use_external_decoder = false;
Configure(audio_config);
RtpCastHeader rtp_header;
rtp_header.webrtc.header.payloadType = 127;
rtp_header.webrtc.header.sequenceNumber = 1234;
rtp_header.webrtc.header.timestamp = 0x87654321;
rtp_header.webrtc.header.ssrc = 0x12345678;
rtp_header.webrtc.header.paddingLength = 0;
rtp_header.webrtc.header.headerLength = 12;
rtp_header.webrtc.type.Audio.isCNG = false;
rtp_header.webrtc.type.Audio.channel = 2;
std::vector<int16> payload(640, 0x1234);
uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
size_t payload_size = payload.size() * sizeof(int16);
audio_decoder_->IncomingParsedRtpPacket(payload_data,
payload_size, rtp_header);
int number_of_10ms_blocks = 2;
int desired_frequency = 16000;
PcmAudioFrame audio_frame;
uint32 rtp_timestamp;
EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
desired_frequency,
&audio_frame,
&rtp_timestamp));
EXPECT_EQ(2, audio_frame.channels);
EXPECT_EQ(16000, audio_frame.frequency);
EXPECT_EQ(640ul, audio_frame.samples.size());
// First 10 samples per channel are 0 from NetEq.
for (size_t i = 10 * audio_config.channels; i < audio_frame.samples.size();
++i) {
EXPECT_EQ(0x3412, audio_frame.samples[i]);
}
rtp_header.webrtc.header.sequenceNumber++;
rtp_header.webrtc.header.timestamp += (audio_config.frequency / 100) * 2 * 2;
audio_decoder_->IncomingParsedRtpPacket(payload_data,
payload_size, rtp_header);
EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
desired_frequency,
&audio_frame,
&rtp_timestamp));
EXPECT_EQ(2, audio_frame.channels);
EXPECT_EQ(16000, audio_frame.frequency);
EXPECT_EQ(640ul, audio_frame.samples.size());
for (size_t i = 0; i < audio_frame.samples.size(); ++i) {
EXPECT_NEAR(0x3412, audio_frame.samples[i], 1000);
}
}
TEST_F(AudioDecoderTest, Pcm16Resample) {
AudioReceiverConfig audio_config;
audio_config.rtp_payload_type = 127;
audio_config.frequency = 16000;
audio_config.channels = 2;
audio_config.codec = kPcm16;
audio_config.use_external_decoder = false;
Configure(audio_config);
RtpCastHeader rtp_header;
rtp_header.webrtc.header.payloadType = 127;
rtp_header.webrtc.header.sequenceNumber = 1234;
rtp_header.webrtc.header.timestamp = 0x87654321;
rtp_header.webrtc.header.ssrc = 0x12345678;
rtp_header.webrtc.header.paddingLength = 0;
rtp_header.webrtc.header.headerLength = 12;
rtp_header.webrtc.type.Audio.isCNG = false;
rtp_header.webrtc.type.Audio.channel = 2;
std::vector<int16> payload(640, 0x1234);
uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
size_t payload_size = payload.size() * sizeof(int16);
audio_decoder_->IncomingParsedRtpPacket(payload_data,
payload_size, rtp_header);
int number_of_10ms_blocks = 2;
int desired_frequency = 48000;
PcmAudioFrame audio_frame;
uint32 rtp_timestamp;
EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
desired_frequency,
&audio_frame,
&rtp_timestamp));
EXPECT_EQ(2, audio_frame.channels);
EXPECT_EQ(48000, audio_frame.frequency);
EXPECT_EQ(1920ul, audio_frame.samples.size()); // Upsampled to 48 KHz.
int count = 0;
// Resampling makes the variance worse.
for (size_t i = 100 * audio_config.channels; i < audio_frame.samples.size();
++i) {
EXPECT_NEAR(0x3412, audio_frame.samples[i], 400);
if (0x3412 == audio_frame.samples[i]) count++;
}
}
} // namespace cast
} // namespace media
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