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AudioDecoder: New method SampleRateHz, + implementations for our codecs

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This will let NetEq (and the factory, and every layer in between) keep
track of just the decoder, instead of decoder and sample rate.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2024633002
Cr-Commit-Position: refs/heads/master@{#12968}
This commit is contained in:
kwiberg
2016-05-31 02:46:20 -07:00
committed by Commit bot
parent 946f36ef39
commit 6c2eab34f8
16 changed files with 125 additions and 27 deletions
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4
webrtc/modules/audio_coding/codecs/audio_decoder.cc
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@ -82,6 +82,10 @@ bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
return false;
}
int AudioDecoder::SampleRateHz() const {
return -1;
}
AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
switch (type) {
case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.

8
webrtc/modules/audio_coding/codecs/audio_decoder.h
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@ -93,6 +93,14 @@ class AudioDecoder {
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
// Returns the actual sample rate of the decoder's output.
// NOTE: For now, this has a default implementation that returns an unusable
// value (-1). That default implementation will go away soon, and at the same
// time callers will start relying on the return value, so make sure you
// override it with something that returns a correct value!
// TODO(kwiberg): Remove the default implementation.
virtual int SampleRateHz() const;
virtual size_t Channels() const = 0;
protected:

17
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
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@ -74,7 +74,7 @@ NamedDecoderConstructor decoder_constructors[] = {
{"isac",
[](const SdpAudioFormat& format) {
return format.clockrate_hz == 16000 && format.num_channels == 1
? Unique(new AudioDecoderIsacFix)
? Unique(new AudioDecoderIsacFix(format.clockrate_hz))
: nullptr;
}},
#elif defined(WEBRTC_CODEC_ISAC)
@ -82,14 +82,15 @@ NamedDecoderConstructor decoder_constructors[] = {
[](const SdpAudioFormat& format) {
return (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
format.num_channels == 1
? Unique(new AudioDecoderIsac)
? Unique(new AudioDecoderIsac(format.clockrate_hz))
: nullptr;
}},
#endif
{"l16",
[](const SdpAudioFormat& format) {
return format.num_channels >= 1
? Unique(new AudioDecoderPcm16B(format.num_channels))
? Unique(new AudioDecoderPcm16B(format.clockrate_hz,
format.num_channels))
: nullptr;
}},
#ifdef WEBRTC_CODEC_G722
@ -136,7 +137,15 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
const SdpAudioFormat& format) override {
for (const auto& dc : decoder_constructors) {
if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
return std::unique_ptr<AudioDecoder>(dc.constructor(format));
std::unique_ptr<AudioDecoder> dec = dc.constructor(format);
if (dec) {
const int expected_sample_rate_hz =
STR_CASE_CMP(format.name.c_str(), "g722") == 0
? 2 * format.clockrate_hz
: format.clockrate_hz;
RTC_CHECK_EQ(expected_sample_rate_hz, dec->SampleRateHz());
}
return dec;
}
}
return nullptr;

12
webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
View File

@ -16,6 +16,10 @@ namespace webrtc {
void AudioDecoderPcmU::Reset() {}
int AudioDecoderPcmU::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderPcmU::Channels() const {
return num_channels_;
}
@ -25,7 +29,7 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
@ -40,6 +44,10 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
void AudioDecoderPcmA::Reset() {}
int AudioDecoderPcmA::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderPcmA::Channels() const {
return num_channels_;
}
@ -49,7 +57,7 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);

2
webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
View File

@ -24,6 +24,7 @@ class AudioDecoderPcmU final : public AudioDecoder {
}
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
@ -45,6 +46,7 @@ class AudioDecoderPcmA final : public AudioDecoder {
}
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:

12
webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
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@ -35,7 +35,7 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 16000);
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
size_t ret =
WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
@ -53,6 +53,10 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
return static_cast<int>(2 * encoded_len / Channels());
}
int AudioDecoderG722::SampleRateHz() const {
return 16000;
}
size_t AudioDecoderG722::Channels() const {
return 1;
}
@ -74,7 +78,7 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 16000);
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
// De-interleave the bit-stream into two separate payloads.
uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
@ -100,6 +104,10 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
return static_cast<int>(ret);
}
int AudioDecoderG722Stereo::SampleRateHz() const {
return 16000;
}
size_t AudioDecoderG722Stereo::Channels() const {
return 2;
}

4
webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
View File

@ -25,6 +25,7 @@ class AudioDecoderG722 final : public AudioDecoder {
bool HasDecodePlc() const override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
@ -44,6 +45,8 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
AudioDecoderG722Stereo();
~AudioDecoderG722Stereo() override;
void Reset() override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
int DecodeInternal(const uint8_t* encoded,
@ -51,7 +54,6 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
size_t Channels() const override;
private:
// Splits the stereo-interleaved payload in |encoded| into separate payloads

4
webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
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@ -49,6 +49,10 @@ void AudioDecoderIlbc::Reset() {
WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
int AudioDecoderIlbc::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderIlbc::Channels() const {
return 1;
}

1
webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
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@ -25,6 +25,7 @@ class AudioDecoderIlbc final : public AudioDecoder {
bool HasDecodePlc() const override;
size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
void Reset() override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:

12
webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
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@ -14,18 +14,24 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
namespace webrtc {
// TODO(kwiberg): Remove the possibility of not specifying the sample rate at
// object creation time.
template <typename T>
class AudioDecoderIsacT final : public AudioDecoder {
public:
AudioDecoderIsacT();
explicit AudioDecoderIsacT(
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
explicit AudioDecoderIsacT(int sample_rate_hz);
AudioDecoderIsacT(int sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
~AudioDecoderIsacT() override;
bool HasDecodePlc() const override;
@ -37,6 +43,7 @@ class AudioDecoderIsacT final : public AudioDecoder {
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) override;
int ErrorCode() override;
int SampleRateHz() const override;
size_t Channels() const override;
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
@ -45,9 +52,12 @@ class AudioDecoderIsacT final : public AudioDecoder {
SpeechType* speech_type) override;
private:
AudioDecoderIsacT(rtc::Optional<int> sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
typename T::instance_type* isac_state_;
rtc::Optional<int> sample_rate_hz_;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
int decoder_sample_rate_hz_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
};

44
webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
View File

@ -19,12 +19,31 @@ namespace webrtc {
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT()
: AudioDecoderIsacT(nullptr) {}
: AudioDecoderIsacT(rtc::Optional<int>(), nullptr) {}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
: bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
: AudioDecoderIsacT(rtc::Optional<int>(), bwinfo) {}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(int sample_rate_hz)
: AudioDecoderIsacT(rtc::Optional<int>(sample_rate_hz), nullptr) {}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(
int sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
: AudioDecoderIsacT(rtc::Optional<int>(sample_rate_hz), bwinfo) {}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(
rtc::Optional<int> sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
: sample_rate_hz_(sample_rate_hz), bwinfo_(bwinfo) {
RTC_CHECK(!sample_rate_hz || *sample_rate_hz == 16000 ||
*sample_rate_hz == 32000)
<< "Unsupported sample rate " << *sample_rate_hz;
RTC_CHECK_EQ(0, T::Create(&isac_state_));
T::DecoderInit(isac_state_);
if (bwinfo_) {
@ -32,6 +51,9 @@ AudioDecoderIsacT<T>::AudioDecoderIsacT(
T::GetBandwidthInfo(isac_state_, &bi);
bwinfo_->Set(bi);
}
if (sample_rate_hz_) {
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, *sample_rate_hz_));
}
}
template <typename T>
@ -45,11 +67,13 @@ int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
<< "Unsupported sample rate " << sample_rate_hz;
if (sample_rate_hz != decoder_sample_rate_hz_) {
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
decoder_sample_rate_hz_ = sample_rate_hz;
if (sample_rate_hz_) {
RTC_CHECK_EQ(*sample_rate_hz_, sample_rate_hz);
} else {
RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
<< "Unsupported sample rate " << sample_rate_hz;
sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz);
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, *sample_rate_hz_));
}
int16_t temp_type = 1; // Default is speech.
int ret =
@ -95,6 +119,12 @@ int AudioDecoderIsacT<T>::ErrorCode() {
return T::GetErrorCode(isac_state_);
}
template <typename T>
int AudioDecoderIsacT<T>::SampleRateHz() const {
RTC_CHECK(sample_rate_hz_) << "Sample rate not set yet!";
return *sample_rate_hz_;
}
template <typename T>
size_t AudioDecoderIsacT<T>::Channels() const {
return 1;

4
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
View File

@ -87,6 +87,10 @@ bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
return (fec == 1);
}
int AudioDecoderOpus::SampleRateHz() const {
return 48000;
}
size_t AudioDecoderOpus::Channels() const {
return channels_;
}

1
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
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@ -27,6 +27,7 @@ class AudioDecoderOpus final : public AudioDecoder {
int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const override;
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:

15
webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
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@ -15,13 +15,20 @@
namespace webrtc {
AudioDecoderPcm16B::AudioDecoderPcm16B(size_t num_channels)
: num_channels_(num_channels) {
AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
: sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_DCHECK_GE(num_channels, 1u);
}
void AudioDecoderPcm16B::Reset() {}
int AudioDecoderPcm16B::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioDecoderPcm16B::Channels() const {
return num_channels_;
}
@ -31,9 +38,7 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
*speech_type = ConvertSpeechType(1);
return static_cast<int>(ret);

4
webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
View File

@ -18,9 +18,10 @@ namespace webrtc {
class AudioDecoderPcm16B final : public AudioDecoder {
public:
explicit AudioDecoderPcm16B(size_t num_channels);
AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
@ -31,6 +32,7 @@ class AudioDecoderPcm16B final : public AudioDecoder {
SpeechType* speech_type) override;
private:
const int sample_rate_hz_;
const size_t num_channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
};