Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call"
This change uncovered issue 4143, evading the Memcheck suppression
since the signature is changed in the Decode function.
A fix for this is in the making; see
https://review.webrtc.org/36309004. This CL will be re-landed once the
fix is in place.
BUG=4143
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42089004
Cr-Commit-Position: refs/heads/master@{#8488}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8488 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 0817873..bc6b9c5 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -18,10 +18,9 @@
int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
- return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
+ return Decode(encoded, encoded_len, decoded, speech_type);
}
bool AudioDecoder::HasDecodePlc() const { return false; }
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 8c83e61..30fdc44 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -37,22 +37,14 @@
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The number of samples from all channels produced is in
// the return value. If the decoder produced comfort noise, |speech_type|
- // is set to kComfortNoise, otherwise it is kSpeech. The desired output
- // sample rate is provided in |sample_rate_hz|, which must be valid for the
- // codec at hand.
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) = 0;
+ // is set to kComfortNoise, otherwise it is kSpeech.
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) = 0;
// Same as Decode(), but interfaces to the decoders redundant decode function.
// The default implementation simply calls the regular Decode() method.
- virtual int DecodeRedundant(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
// Indicates if the decoder implements the DecodePlc method.
virtual bool HasDecodePlc() const;
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 668f491..994d21b 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
@@ -66,6 +66,8 @@
explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
virtual ~AudioEncoderDecoderIsacT() OVERRIDE;
+ void UpdateDecoderSampleRate(int sample_rate_hz);
+
// AudioEncoder public methods.
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
@@ -75,12 +77,10 @@
// AudioDecoder methods.
virtual int Decode(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual bool HasDecodePlc() const OVERRIDE;
@@ -116,8 +116,6 @@
typename T::instance_type* isac_state_
GUARDED_BY(state_lock_) /* PT_GUARDED_BY(lock_)*/;
- int decoder_sample_rate_hz_ GUARDED_BY(state_lock_);
-
// Must be acquired before state_lock_.
const scoped_ptr<CriticalSectionWrapper> lock_;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index d9cec82..095bb7b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -109,7 +109,6 @@
: payload_type_(config.payload_type),
red_payload_type_(config.red_payload_type),
state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- decoder_sample_rate_hz_(0),
lock_(CriticalSectionWrapper::CreateCriticalSection()),
packet_in_progress_(false),
redundant_length_bytes_(0) {
@@ -137,7 +136,6 @@
: payload_type_(config.payload_type),
red_payload_type_(config.red_payload_type),
state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
- decoder_sample_rate_hz_(0),
lock_(CriticalSectionWrapper::CreateCriticalSection()),
packet_in_progress_(false),
redundant_length_bytes_(0) {
@@ -162,6 +160,12 @@
}
template <typename T>
+void AudioEncoderDecoderIsacT<T>::UpdateDecoderSampleRate(int sample_rate_hz) {
+ CriticalSectionScoped cs(state_lock_.get());
+ CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
+}
+
+template <typename T>
int AudioEncoderDecoderIsacT<T>::SampleRateHz() const {
CriticalSectionScoped cs(state_lock_.get());
return T::EncSampRate(isac_state_);
@@ -266,16 +270,9 @@
template <typename T>
int AudioEncoderDecoderIsacT<T>::Decode(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped cs(state_lock_.get());
- CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
- << "Unsupported sample rate " << sample_rate_hz;
- if (sample_rate_hz != decoder_sample_rate_hz_) {
- CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
- decoder_sample_rate_hz_ = sample_rate_hz;
- }
int16_t temp_type = 1; // Default is speech.
int16_t ret =
T::Decode(isac_state_, encoded, static_cast<int16_t>(encoded_len),
@@ -287,7 +284,6 @@
template <typename T>
int AudioEncoderDecoderIsacT<T>::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
- int /*sample_rate_hz*/,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped cs(state_lock_.get());
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 71df8a3..1a0e483 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -158,22 +158,18 @@
int AudioDecoderProxy::Decode(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
- return decoder_->Decode(encoded, encoded_len, sample_rate_hz, decoded,
- speech_type);
+ return decoder_->Decode(encoded, encoded_len, decoded, speech_type);
}
int AudioDecoderProxy::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
- return decoder_->DecodeRedundant(encoded, encoded_len, sample_rate_hz,
- decoded, speech_type);
+ return decoder_->DecodeRedundant(encoded, encoded_len, decoded, speech_type);
}
bool AudioDecoderProxy::HasDecodePlc() const {
@@ -553,6 +549,28 @@
ResetAudioEncoder();
}
+int16_t ACMGenericCodec::UpdateDecoderSampFreq(int16_t codec_id) {
+#ifdef WEBRTC_CODEC_ISAC
+ WriteLockScoped wl(codec_wrapper_lock_);
+ if (is_isac_) {
+ switch (codec_id) {
+ case ACMCodecDB::kISAC:
+ static_cast<AudioEncoderDecoderIsac*>(audio_encoder_.get())
+ ->UpdateDecoderSampleRate(16000);
+ return 0;
+ case ACMCodecDB::kISACSWB:
+ case ACMCodecDB::kISACFB:
+ static_cast<AudioEncoderDecoderIsac*>(audio_encoder_.get())
+ ->UpdateDecoderSampleRate(32000);
+ return 0;
+ default:
+ FATAL() << "Unexpected codec id.";
+ }
+ }
+#endif
+ return 0;
+}
+
int32_t ACMGenericCodec::SetISACMaxPayloadSize(
const uint16_t max_payload_len_bytes) {
WriteLockScoped wl(codec_wrapper_lock_);
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 fd3fbab..933306d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
@@ -48,12 +48,10 @@
bool IsSet() const;
int Decode(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
- int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
bool HasDecodePlc() const override;
@@ -296,6 +294,33 @@
void SetCngPt(int sample_rate_hz, int payload_type);
///////////////////////////////////////////////////////////////////////////
+ // UpdateDecoderSampFreq()
+ // For most of the codecs this function does nothing. It must be
+ // implemented for those codecs that one codec instance serves as the
+ // decoder for different flavors of the codec. One example is iSAC. there,
+ // iSAC 16 kHz and iSAC 32 kHz are treated as two different codecs with
+ // different payload types, however, there is only one iSAC instance to
+ // decode. The reason for that is we would like to decode and encode with
+ // the same codec instance for bandwidth estimator to work.
+ //
+ // Each time that we receive a new payload type, we call this function to
+ // prepare the decoder associated with the new payload. Normally, decoders
+ // doesn't have to do anything. For iSAC the decoder has to change it's
+ // sampling rate. The input parameter specifies the current flavor of the
+ // codec in codec database. For instance, if we just got a SWB payload then
+ // the input parameter is ACMCodecDB::isacswb.
+ //
+ // Input:
+ // -codec_id : the ID of the codec associated with the
+ // payload type that we just received.
+ //
+ // Return value:
+ // 0 if succeeded in updating the decoder.
+ // -1 if failed to update.
+ //
+ int16_t UpdateDecoderSampFreq(int16_t /* codec_id */);
+
+ ///////////////////////////////////////////////////////////////////////////
// UpdateEncoderSampFreq()
// Call this function to update the encoder sampling frequency. This
// is for codecs where one payload-name supports several encoder sampling
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 5041eb7..7acb45a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -609,6 +609,14 @@
return last_audio_decoder_;
}
+int AcmReceiver::last_audio_payload_type() const {
+ CriticalSectionScoped lock(crit_sect_.get());
+ if (last_audio_decoder_ < 0)
+ return -1;
+ assert(decoders_[last_audio_decoder_].registered);
+ return decoders_[last_audio_decoder_].payload_type;
+}
+
int AcmReceiver::RedPayloadType() const {
CriticalSectionScoped lock(crit_sect_.get());
if (ACMCodecDB::kRED < 0 ||
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 43f304a..f3ef16f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -248,6 +248,12 @@
int last_audio_codec_id() const; // TODO(turajs): can be inline.
//
+ // Return the payload-type of the last non-CNG/non-DTMF RTP packet. If no
+ // non-CNG/non-DTMF packet is received -1 is returned.
+ //
+ int last_audio_payload_type() const; // TODO(turajs): can be inline.
+
+ //
// Get the audio codec associated with the last non-CNG/non-DTMF received
// payload. If no non-CNG/non-DTMF packet is received -1 is returned,
// otherwise return 0.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 6b15718..273bb5a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -321,6 +321,7 @@
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
EXPECT_EQ(-1, receiver_->last_audio_codec_id());
+ EXPECT_EQ(-1, receiver_->last_audio_payload_type());
n = 0;
while (kCodecId[n] >= 0) { // Loop over codecs.
@@ -346,6 +347,8 @@
ASSERT_TRUE(packet_sent_);
}
EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
+ EXPECT_EQ(codecs_[kCodecId[n]].pltype,
+ receiver_->last_audio_payload_type());
EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec));
++n;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index d57d511..5b2c3cb 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -325,6 +325,7 @@
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
EXPECT_EQ(-1, receiver_->last_audio_codec_id());
+ EXPECT_EQ(-1, receiver_->last_audio_payload_type());
n = 0;
while (kCodecId[n] >= 0) { // Loop over codecs.
@@ -350,6 +351,8 @@
ASSERT_TRUE(packet_sent_);
}
EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
+ EXPECT_EQ(codecs_[kCodecId[n]].pltype,
+ receiver_->last_audio_payload_type());
EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec));
++n;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index bf9bb01..41e8d4d 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -1185,7 +1185,24 @@
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
const WebRtcRTPHeader& rtp_header) {
- return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length);
+ int last_audio_pltype = receiver_.last_audio_payload_type();
+ if (receiver_.InsertPacket(rtp_header, incoming_payload, payload_length) <
+ 0) {
+ return -1;
+ }
+ if (receiver_.last_audio_payload_type() != last_audio_pltype) {
+ int index = receiver_.last_audio_codec_id();
+ assert(index >= 0);
+ CriticalSectionScoped lock(acm_crit_sect_);
+
+ // |codec_[index]| might not be even created, simply because it is not
+ // yet registered as send codec. Even if it is registered, unless the
+ // codec shares same instance for encoder and decoder, this call is
+ // useless.
+ if (codecs_[index] != NULL)
+ codecs_[index]->UpdateDecoderSampFreq(index);
+ }
+ return 0;
}
// Minimum playout delay (Used for lip-sync).
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index f77dead..43ba241 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -38,12 +38,8 @@
namespace webrtc {
// PCMu
-int AudioDecoderPcmU::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 8000);
+int AudioDecoderPcmU::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@@ -58,12 +54,8 @@
}
// PCMa
-int AudioDecoderPcmA::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 8000);
+int AudioDecoderPcmA::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@@ -81,14 +73,8 @@
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}
-int AudioDecoderPcm16B::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
- sample_rate_hz == 32000 || sample_rate_hz == 48000)
- << "Unsupported sample rate " << sample_rate_hz;
+int AudioDecoderPcm16B::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t ret =
WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded);
*speech_type = ConvertSpeechType(1);
@@ -117,12 +103,8 @@
WebRtcIlbcfix_DecoderFree(dec_state_);
}
-int AudioDecoderIlbc::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 8000);
+int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@@ -150,12 +132,8 @@
WebRtcG722_FreeDecoder(dec_state_);
}
-int AudioDecoderG722::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 16000);
+int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret =
WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
@@ -185,12 +163,8 @@
WebRtcG722_FreeDecoder(dec_state_right_);
}
-int AudioDecoderG722Stereo::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 16000);
+int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
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];
@@ -270,12 +244,8 @@
WebRtcOpus_DecoderFree(dec_state_);
}
-int AudioDecoderOpus::Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
- DCHECK_EQ(sample_rate_hz, 48000);
+int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@@ -287,13 +257,11 @@
}
int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
+ size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
- return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
+ return Decode(encoded, encoded_len, decoded, speech_type);
}
int16_t temp_type = 1; // Default is speech.
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 7d36a39..57bd522 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -37,11 +37,8 @@
class AudioDecoderPcmU : public AudioDecoder {
public:
AudioDecoderPcmU() {}
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -52,11 +49,8 @@
class AudioDecoderPcmA : public AudioDecoder {
public:
AudioDecoderPcmA() {}
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -92,11 +86,8 @@
class AudioDecoderPcm16B : public AudioDecoder {
public:
AudioDecoderPcm16B();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -121,11 +112,8 @@
public:
AudioDecoderIlbc();
virtual ~AudioDecoderIlbc();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual bool HasDecodePlc() const { return true; }
virtual int DecodePlc(int num_frames, int16_t* decoded);
virtual int Init();
@@ -141,11 +129,8 @@
public:
AudioDecoderG722();
virtual ~AudioDecoderG722();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual bool HasDecodePlc() const { return false; }
virtual int Init();
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -159,11 +144,8 @@
public:
AudioDecoderG722Stereo();
virtual ~AudioDecoderG722Stereo();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual int Init();
private:
@@ -187,16 +169,10 @@
public:
explicit AudioDecoderOpus(int num_channels);
virtual ~AudioDecoderOpus();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
- virtual int DecodeRedundant(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
+ virtual int DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type);
virtual int Init();
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
virtual int PacketDurationRedundant(const uint8_t* encoded,
@@ -219,13 +195,8 @@
public:
explicit AudioDecoderCng();
virtual ~AudioDecoderCng();
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int /*sample_rate_hz*/,
- int16_t* decoded,
- SpeechType* speech_type) {
- return -1;
- }
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) { return -1; }
virtual int Init();
virtual int IncomingPacket(const uint8_t* payload,
size_t payload_len,
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 1f0e881..95805d3 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -186,9 +186,10 @@
// Make sure that frame_size_ * channels_ samples are allocated and free.
decoded.resize((processed_samples + frame_size_) * channels_, 0);
AudioDecoder::SpeechType speech_type;
- size_t dec_len = decoder_->Decode(
- &encoded_[encoded_bytes_], enc_len, codec_input_rate_hz_,
- &decoded[processed_samples * channels_], &speech_type);
+ size_t dec_len = decoder_->Decode(&encoded_[encoded_bytes_],
+ enc_len,
+ &decoded[processed_samples * channels_],
+ &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
encoded_bytes_ += enc_len;
processed_samples += frame_size_;
@@ -221,15 +222,13 @@
AudioDecoder::SpeechType speech_type1, speech_type2;
EXPECT_EQ(0, decoder_->Init());
scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
- dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
- output1.get(), &speech_type1);
+ dec_len = decoder_->Decode(encoded_, enc_len, output1.get(), &speech_type1);
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Re-init decoder and decode again.
EXPECT_EQ(0, decoder_->Init());
scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
- dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
- output2.get(), &speech_type2);
+ dec_len = decoder_->Decode(encoded_, enc_len, output2.get(), &speech_type2);
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
for (unsigned int n = 0; n < frame_size_; ++n) {
@@ -248,8 +247,8 @@
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
- size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
- output.get(), &speech_type);
+ size_t dec_len =
+ decoder_->Decode(encoded_, enc_len, output.get(), &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Call DecodePlc and verify that we get one frame of data.
// (Overwrite the output from the above Decode call, but that does not
@@ -339,8 +338,8 @@
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(0, decoder_->Init());
scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
- size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
- output.get(), &speech_type);
+ size_t dec_len =
+ decoder_->Decode(encoded_, enc_len, output.get(), &speech_type);
EXPECT_EQ(frame_size_, dec_len);
// Simply call DecodePlc and verify that we get 0 as return value.
EXPECT_EQ(0, decoder_->DecodePlc(1, output.get()));
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 7288f11..503e46f 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -22,9 +22,8 @@
MockAudioDecoder() {}
virtual ~MockAudioDecoder() { Die(); }
MOCK_METHOD0(Die, void());
- MOCK_METHOD5(
- Decode,
- int(const uint8_t*, size_t, int, int16_t*, AudioDecoder::SpeechType*));
+ MOCK_METHOD4(Decode, int(const uint8_t*, size_t, int16_t*,
+ AudioDecoder::SpeechType*));
MOCK_CONST_METHOD0(HasDecodePlc, bool());
MOCK_METHOD2(DecodePlc, int(int, int16_t*));
MOCK_METHOD0(Init, int());
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 22d2816..19f069a 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
@@ -29,11 +29,8 @@
public:
ExternalPcm16B() {}
- virtual int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) {
+ virtual int Decode(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) {
int16_t ret = WebRtcPcm16b_Decode(
encoded, static_cast<int16_t>(encoded_len), decoded);
*speech_type = ConvertSpeechType(1);
@@ -52,7 +49,7 @@
public:
MockExternalPcm16B() {
// By default, all calls are delegated to the real object.
- ON_CALL(*this, Decode(_, _, _, _, _))
+ ON_CALL(*this, Decode(_, _, _, _))
.WillByDefault(Invoke(&real_, &ExternalPcm16B::Decode));
ON_CALL(*this, HasDecodePlc())
.WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
@@ -68,12 +65,9 @@
virtual ~MockExternalPcm16B() { Die(); }
MOCK_METHOD0(Die, void());
- MOCK_METHOD5(Decode,
- int(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type));
+ MOCK_METHOD4(Decode,
+ int(const uint8_t* encoded, size_t encoded_len, int16_t* decoded,
+ SpeechType* speech_type));
MOCK_CONST_METHOD0(HasDecodePlc,
bool());
MOCK_METHOD2(DecodePlc,
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 0449044..a3dd271 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -100,8 +100,7 @@
next_arrival_time = GetArrivalTime(next_send_time);
} while (Lost()); // If lost, immediately read the next packet.
- EXPECT_CALL(*external_decoder_,
- Decode(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _))
+ EXPECT_CALL(*external_decoder_, Decode(_, payload_size_bytes_, _, _))
.Times(NumExpectedDecodeCalls(num_loops));
uint32_t time_now = 0;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 7370825..f1a3a90 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -1266,7 +1266,7 @@
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
decode_length = decoder->DecodeRedundant(
- packet->payload, packet->payload_length, fs_hz_,
+ packet->payload, packet->payload_length,
&decoded_buffer_[*decoded_length], speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
@@ -1274,9 +1274,10 @@
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
- decode_length =
- decoder->Decode(packet->payload, packet->payload_length, fs_hz_,
- &decoded_buffer_[*decoded_length], speech_type);
+ decode_length = decoder->Decode(packet->payload,
+ packet->payload_length,
+ &decoded_buffer_[*decoded_length],
+ speech_type);
}
delete[] packet->payload;
@@ -1606,8 +1607,7 @@
if (decoder) {
const uint8_t* dummy_payload = NULL;
AudioDecoder::SpeechType speech_type;
- length =
- decoder->Decode(dummy_payload, 0, fs_hz_, decoded_buffer, &speech_type);
+ length = decoder->Decode(dummy_payload, 0, decoded_buffer, &speech_type);
}
assert(mute_factor_array_.get());
normal_->Process(decoded_buffer, length, last_mode_, mute_factor_array_.get(),
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 54b393b..36ed35a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -430,7 +430,6 @@
// Produce as many samples as input bytes (|encoded_len|).
virtual int Decode(const uint8_t* encoded,
size_t encoded_len,
- int /*sample_rate_hz*/,
int16_t* decoded,
SpeechType* speech_type) {
for (size_t i = 0; i < encoded_len; ++i) {
@@ -522,11 +521,10 @@
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
// |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
- EXPECT_CALL(mock_decoder,
- Decode(Pointee(0), kPayloadLengthBytes, kSampleRateHz, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
+ SetArgPointee<3>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterExternalDecoder(
@@ -568,11 +566,10 @@
// Expect only the second packet to be decoded (the one with "2" as the first
// payload byte).
- EXPECT_CALL(mock_decoder,
- Decode(Pointee(2), kPayloadLengthBytes, kSampleRateHz, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
+ SetArgPointee<3>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
// Pull audio once.
@@ -685,31 +682,28 @@
// Pointee(x) verifies that first byte of the payload equals x, this makes it
// possible to verify that the correct payload is fed to Decode().
- EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
+ SetArgPointee<3>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kComfortNoise),
+ SetArgPointee<3>(AudioDecoder::kComfortNoise),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kComfortNoise),
+ SetArgPointee<3>(AudioDecoder::kComfortNoise),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes,
- kSampleRateKhz * 1000, _, _))
- .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes, _, _))
+ .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<4>(AudioDecoder::kSpeech),
+ SetArgPointee<3>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
EXPECT_EQ(NetEq::kOK,
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 cdcf0b3..b61bf83 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -36,22 +36,16 @@
MOCK_METHOD0(Init, int());
// Override the following methods such that no actual payload is needed.
- int Decode(const uint8_t* encoded,
- size_t encoded_len,
- int /*sample_rate_hz*/,
- int16_t* decoded,
+ int Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) override {
*speech_type = kSpeech;
memset(decoded, 0, sizeof(int16_t) * kPacketDuration * channels_);
return kPacketDuration * channels_;
}
- int DecodeRedundant(const uint8_t* encoded,
- size_t encoded_len,
- int sample_rate_hz,
- int16_t* decoded,
- SpeechType* speech_type) override {
- return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
+ int DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
+ int16_t* decoded, SpeechType* speech_type) override {
+ return Decode(encoded, encoded_len, decoded, speech_type);
}
int PacketDuration(const uint8_t* encoded,
