Make AudioDecoder stateless
The channels_ member varable is removed from the base class, and the
associated accessor function is changed to Channels() which is a pure
virtual function.
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43779004
Cr-Commit-Position: refs/heads/master@{#8775}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8775 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 22e44a4..1ac02c5 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -31,8 +31,8 @@
// Used by PacketDuration below. Save the value -1 for errors.
enum { kNotImplemented = -2 };
- AudioDecoder() : channels_(1) {}
- virtual ~AudioDecoder() {}
+ AudioDecoder() = default;
+ virtual ~AudioDecoder() = default;
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The maximum bytes allowed to be written into |decoded| is
@@ -97,7 +97,7 @@
// isn't a CNG decoder, don't call this method.
virtual CNG_dec_inst* CngDecoderInstance();
- size_t channels() const { return channels_; }
+ virtual size_t Channels() const = 0;
protected:
static SpeechType ConvertSpeechType(int16_t type);
@@ -114,8 +114,6 @@
int16_t* decoded,
SpeechType* speech_type);
- size_t channels_;
-
private:
DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
};
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 6b197bc..077568a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -81,6 +81,7 @@
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) override;
int ErrorCode() override;
+ size_t Channels() const override { return 1; }
protected:
// AudioEncoder protected method.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
index cf407db..2fc4306 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -121,7 +121,6 @@
void AudioDecoderProxy::SetDecoder(AudioDecoder* decoder) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
decoder_ = decoder;
- channels_ = decoder->channels();
CHECK_EQ(decoder_->Init(), 0);
}
@@ -205,6 +204,11 @@
return decoder_->CngDecoderInstance();
}
+size_t AudioDecoderProxy::Channels() const {
+ CriticalSectionScoped decoder_lock(decoder_lock_.get());
+ return decoder_->Channels();
+}
+
int16_t ACMGenericCodec::EncoderParams(WebRtcACMCodecParams* enc_params) {
*enc_params = acm_codec_params_;
return 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
index 93fda43..cf07ea4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
@@ -73,6 +73,7 @@
size_t encoded_len) const override;
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
CNG_dec_inst* CngDecoderInstance() override;
+ size_t Channels() const override;
private:
rtc::scoped_ptr<CriticalSectionWrapper> decoder_lock_;
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 8ffb761..ce24e4a 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -54,7 +54,7 @@
int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
- return static_cast<int>(encoded_len / channels_);
+ return static_cast<int>(encoded_len / Channels());
}
// PCMa
@@ -74,7 +74,7 @@
int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
- return static_cast<int>(encoded_len / channels_);
+ return static_cast<int>(encoded_len / Channels());
}
// PCM16B
@@ -98,12 +98,12 @@
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// Two encoded byte per sample per channel.
- return static_cast<int>(encoded_len / (2 * channels_));
+ return static_cast<int>(encoded_len / (2 * Channels()));
}
-AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) {
+AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels)
+ : channels_(num_channels) {
DCHECK(num_channels > 0);
- channels_ = num_channels;
}
#endif
@@ -171,11 +171,10 @@
int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// 1/2 encoded byte per sample per channel.
- return static_cast<int>(2 * encoded_len / channels_);
+ return static_cast<int>(2 * encoded_len / Channels());
}
AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
- channels_ = 2;
WebRtcG722_CreateDecoder(&dec_state_left_);
WebRtcG722_CreateDecoder(&dec_state_right_);
}
@@ -260,9 +259,8 @@
// Opus
#ifdef WEBRTC_CODEC_OPUS
-AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
+AudioDecoderOpus::AudioDecoderOpus(int num_channels) : channels_(num_channels) {
DCHECK(num_channels == 1 || num_channels == 2);
- channels_ = num_channels;
WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
}
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 2222b62..5f9c35b 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -39,6 +39,7 @@
AudioDecoderPcmU() {}
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -56,6 +57,7 @@
AudioDecoderPcmA() {}
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -70,23 +72,27 @@
class AudioDecoderPcmUMultiCh : public AudioDecoderPcmU {
public:
- explicit AudioDecoderPcmUMultiCh(size_t channels) : AudioDecoderPcmU() {
+ explicit AudioDecoderPcmUMultiCh(size_t channels)
+ : AudioDecoderPcmU(), channels_(channels) {
assert(channels > 0);
- channels_ = channels;
}
+ size_t Channels() const override { return channels_; }
private:
+ const size_t channels_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmUMultiCh);
};
class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
public:
- explicit AudioDecoderPcmAMultiCh(size_t channels) : AudioDecoderPcmA() {
+ explicit AudioDecoderPcmAMultiCh(size_t channels)
+ : AudioDecoderPcmA(), channels_(channels) {
assert(channels > 0);
- channels_ = channels;
}
+ size_t Channels() const override { return channels_; }
private:
+ const size_t channels_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmAMultiCh);
};
@@ -98,6 +104,7 @@
AudioDecoderPcm16B();
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -116,8 +123,10 @@
class AudioDecoderPcm16BMultiCh : public AudioDecoderPcm16B {
public:
explicit AudioDecoderPcm16BMultiCh(int num_channels);
+ size_t Channels() const override { return channels_; }
private:
+ const size_t channels_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16BMultiCh);
};
#endif
@@ -130,6 +139,7 @@
virtual bool HasDecodePlc() const { return true; }
virtual int DecodePlc(int num_frames, int16_t* decoded);
virtual int Init();
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -152,6 +162,7 @@
virtual bool HasDecodePlc() const { return false; }
virtual int Init();
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -177,6 +188,7 @@
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
+ size_t Channels() const override { return 2; }
private:
// Splits the stereo-interleaved payload in |encoded| into separate payloads
@@ -205,6 +217,7 @@
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
+ size_t Channels() const override { return channels_; }
protected:
int DecodeInternal(const uint8_t* encoded,
@@ -220,6 +233,7 @@
private:
OpusDecInst* dec_state_;
+ const size_t channels_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
};
#endif
@@ -242,6 +256,7 @@
uint32_t arrival_timestamp) { return -1; }
CNG_dec_inst* CngDecoderInstance() override { return dec_state_; }
+ size_t Channels() const override { return 1; }
protected:
int DecodeInternal(const uint8_t* encoded,
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
index b113e4a..93261ab 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -31,6 +31,7 @@
MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
uint32_t));
MOCK_METHOD0(ErrorCode, int());
+ MOCK_CONST_METHOD0(Channels, size_t());
MOCK_CONST_METHOD0(codec_type, NetEqDecoder());
MOCK_METHOD1(CodecSupported, bool(NetEqDecoder));
};
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
index beff6ae6..d8c8856 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
@@ -41,6 +41,7 @@
*speech_type = ConvertSpeechType(1);
return ret;
}
+ size_t Channels() const override { return 1; }
private:
DISALLOW_COPY_AND_ASSIGN(ExternalPcm16B);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index fb9656b..32c6629 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -644,8 +644,8 @@
decoder_database_->GetDecoderInfo(payload_type);
assert(decoder_info);
if (decoder_info->fs_hz != fs_hz_ ||
- decoder->channels() != algorithm_buffer_->Channels())
- SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
+ decoder->Channels() != algorithm_buffer_->Channels())
+ SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
}
// TODO(hlundin): Move this code to DelayManager class.
@@ -1153,9 +1153,9 @@
// If sampling rate or number of channels has changed, we need to make
// a reset.
if (decoder_info->fs_hz != fs_hz_ ||
- decoder->channels() != algorithm_buffer_->Channels()) {
+ decoder->Channels() != algorithm_buffer_->Channels()) {
// TODO(tlegrand): Add unittest to cover this event.
- SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
+ SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
}
sync_buffer_->set_end_timestamp(timestamp_);
playout_timestamp_ = timestamp_;
@@ -1219,7 +1219,7 @@
// since in this case, the we will increment the CNGplayedTS counter.
// Increase with number of samples per channel.
assert(*decoded_length == 0 ||
- (decoder && decoder->channels() == sync_buffer_->Channels()));
+ (decoder && decoder->Channels() == sync_buffer_->Channels()));
sync_buffer_->IncreaseEndTimestamp(
*decoded_length / static_cast<int>(sync_buffer_->Channels()));
}
@@ -1239,8 +1239,8 @@
assert(decoder); // At this point, we must have a decoder object.
// The number of channels in the |sync_buffer_| should be the same as the
// number decoder channels.
- assert(sync_buffer_->Channels() == decoder->channels());
- assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->channels());
+ assert(sync_buffer_->Channels() == decoder->Channels());
+ assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
assert(*operation == kNormal || *operation == kAccelerate ||
*operation == kMerge || *operation == kPreemptiveExpand);
packet_list->pop_front();
@@ -1254,8 +1254,9 @@
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
- memset(&decoded_buffer_[*decoded_length], 0, decoder_frame_length_ *
- decoder->channels() * sizeof(decoded_buffer_[0]));
+ memset(&decoded_buffer_[*decoded_length], 0,
+ decoder_frame_length_ * decoder->Channels() *
+ sizeof(decoded_buffer_[0]));
decode_length = decoder_frame_length_;
} else if (!packet->primary) {
// This is a redundant payload; call the special decoder method.
@@ -1288,11 +1289,11 @@
if (decode_length > 0) {
*decoded_length += decode_length;
// Update |decoder_frame_length_| with number of samples per channel.
- decoder_frame_length_ = decode_length /
- static_cast<int>(decoder->channels());
- LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples (" <<
- decoder->channels() << " channel(s) -> " << decoder_frame_length_ <<
- " samples per channel)";
+ decoder_frame_length_ =
+ decode_length / static_cast<int>(decoder->Channels());
+ LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
+ << decoder->Channels() << " channel(s) -> "
+ << decoder_frame_length_ << " samples per channel)";
} else if (decode_length < 0) {
// Error.
LOG_FERR2(LS_WARNING, Decode, decode_length, payload_length);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 0a5c6a4..3823d96 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -270,6 +270,7 @@
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
+ EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
// BWE update function called with first packet.
EXPECT_CALL(mock_decoder, IncomingPacket(_,
kPayloadLength,
@@ -447,6 +448,8 @@
return 0;
}
+ size_t Channels() const override { return 1; }
+
uint16_t next_value() const { return next_value_; }
private:
@@ -519,6 +522,7 @@
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+ EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
int16_t dummy_output[kPayloadLengthSamples] = {0};
@@ -682,6 +686,7 @@
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+ EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
@@ -824,6 +829,7 @@
MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
SpeechType*));
+ size_t Channels() const override { return 1; }
} decoder_;
const uint8_t kFirstPayloadValue = 1;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index a49f957..e1a0f69 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -61,8 +61,8 @@
int16_t* decoded,
SpeechType* speech_type) override {
*speech_type = kSpeech;
- memset(decoded, 0, sizeof(int16_t) * kPacketDuration * channels_);
- return kPacketDuration * channels_;
+ memset(decoded, 0, sizeof(int16_t) * kPacketDuration * Channels());
+ return kPacketDuration * Channels();
}
int DecodeRedundantInternal(const uint8_t* encoded,
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 17391da..3eb4a29 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -21,7 +21,7 @@
: codec_(codec),
decoder_(decoder),
sample_rate_hz_(CodecSampleRateHz(codec_)),
- channels_(static_cast<int>(decoder_->channels())) {
+ channels_(static_cast<int>(decoder_->Channels())) {
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz_;
neteq_.reset(NetEq::Create(config));
