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AudioDecoderOpus: Add support for 16 kHz output sample rate

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In addition to the 48 kHz that we've always used.

Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
This commit is contained in:
Karl Wiberg
2019-05-29 13:46:09 +02:00
committed by Commit Bot
parent ed69d41b62
commit 7eb0a5e210
5 changed files with 66 additions and 60 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
modules/audio_coding/codecs/opus/audio_decoder_opus.cc
View File

@ -20,10 +20,13 @@
namespace webrtc {
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
: channels_(num_channels) {
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
int sample_rate_hz)
: channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_, 48000);
RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
const int error =
WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
RTC_DCHECK(error == 0);
WebRtcOpus_DecoderInit(dec_state_);
}
@ -57,7 +60,7 @@ int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 48000);
RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
int16_t temp_type = 1; // Default is speech.
int ret =
WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
@ -78,7 +81,7 @@ int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
speech_type);
}
RTC_DCHECK_EQ(sample_rate_hz, 48000);
RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
&temp_type);
@ -104,7 +107,7 @@ int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
return PacketDuration(encoded, encoded_len);
}
return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
}
bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
@ -115,7 +118,7 @@ bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
}
int AudioDecoderOpusImpl::SampleRateHz() const {
return 48000;
return sample_rate_hz_;
}
size_t AudioDecoderOpusImpl::Channels() const {

4
modules/audio_coding/codecs/opus/audio_decoder_opus.h
View File

@ -24,7 +24,8 @@ namespace webrtc {
class AudioDecoderOpusImpl final : public AudioDecoder {
public:
explicit AudioDecoderOpusImpl(size_t num_channels);
explicit AudioDecoderOpusImpl(size_t num_channels,
int sample_rate_hz = 48000);
~AudioDecoderOpusImpl() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
@ -52,6 +53,7 @@ class AudioDecoderOpusImpl final : public AudioDecoder {
private:
OpusDecInst* dec_state_;
const size_t channels_;
const int sample_rate_hz_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpusImpl);
};

80
modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -426,33 +426,34 @@ class AudioDecoderG722StereoTest : public AudioDecoderTest {
}
};
class AudioDecoderOpusTest : public AudioDecoderTest {
class AudioDecoderOpusTest
: public AudioDecoderTest,
public testing::WithParamInterface<std::tuple<int, int>> {
protected:
AudioDecoderOpusTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 48000;
frame_size_ = 480;
channels_ = opus_num_channels_;
codec_input_rate_hz_ = opus_sample_rate_hz_;
frame_size_ = rtc::CheckedDivExact(opus_sample_rate_hz_, 100);
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpusImpl(1);
decoder_ =
new AudioDecoderOpusImpl(opus_num_channels_, opus_sample_rate_hz_);
AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
config.frame_size_ms = 10;
config.sample_rate_hz = opus_sample_rate_hz_;
config.num_channels = opus_num_channels_;
config.application = opus_num_channels_ == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
}
const int opus_sample_rate_hz_{std::get<0>(GetParam())};
const int opus_num_channels_{std::get<1>(GetParam())};
};
class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
protected:
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
delete decoder_;
decoder_ = new AudioDecoderOpusImpl(2);
AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;
config.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
}
};
INSTANTIATE_TEST_SUITE_P(Param,
AudioDecoderOpusTest,
testing::Combine(testing::Values(16000, 48000),
testing::Values(1, 2)));
TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
int tolerance = 251;
@ -592,41 +593,22 @@ TEST_F(AudioDecoderG722StereoTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
}
TEST_F(AudioDecoderOpusTest, EncodeDecode) {
int tolerance = 6176;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
namespace {
void TestOpusSetTargetBitrates(AudioEncoder* audio_encoder) {
EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder, 5999));
EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder, 6000));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder, 32000));
EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder, 510000));
EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder, 511000));
}
} // namespace
TEST_F(AudioDecoderOpusTest, SetTargetBitrate) {
TestOpusSetTargetBitrates(audio_encoder_.get());
}
TEST_F(AudioDecoderOpusStereoTest, EncodeDecode) {
int tolerance = 6176;
int channel_diff_tolerance = 0;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
TEST_P(AudioDecoderOpusTest, EncodeDecode) {
constexpr int tolerance = 6176;
const int channel_diff_tolerance = opus_sample_rate_hz_ == 16000 ? 6 : 0;
constexpr double mse = 238630.0;
constexpr int delay = 22; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderOpusStereoTest, SetTargetBitrate) {
TestOpusSetTargetBitrates(audio_encoder_.get());
TEST_P(AudioDecoderOpusTest, SetTargetBitrate) {
EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder_.get(), 5999));
EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder_.get(), 6000));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(), 32000));
EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(), 510000));
EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(), 511000));
}
} // namespace webrtc