Set decoder output frequency in AudioDecoder::Decode call
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34349004
Cr-Commit-Position: refs/heads/master@{#8476}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 43ba241..f77dead 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -38,8 +38,12 @@
namespace webrtc {
// PCMu
-int AudioDecoderPcmU::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@@ -54,8 +58,12 @@
}
// PCMa
-int AudioDecoderPcmA::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
@@ -73,8 +81,14 @@
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}
-int AudioDecoderPcm16B::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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;
int16_t ret =
WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded);
*speech_type = ConvertSpeechType(1);
@@ -103,8 +117,12 @@
WebRtcIlbcfix_DecoderFree(dec_state_);
}
-int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@@ -132,8 +150,12 @@
WebRtcG722_FreeDecoder(dec_state_);
}
-int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
int16_t temp_type = 1; // Default is speech.
int16_t ret =
WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
@@ -163,8 +185,12 @@
WebRtcG722_FreeDecoder(dec_state_right_);
}
-int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
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];
@@ -244,8 +270,12 @@
WebRtcOpus_DecoderFree(dec_state_);
}
-int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+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);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
@@ -257,11 +287,13 @@
}
int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
- size_t encoded_len, int16_t* decoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
- return Decode(encoded, encoded_len, decoded, speech_type);
+ return Decode(encoded, encoded_len, sample_rate_hz, 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 57bd522..7d36a39 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -37,8 +37,11 @@
class AudioDecoderPcmU : public AudioDecoder {
public:
AudioDecoderPcmU() {}
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -49,8 +52,11 @@
class AudioDecoderPcmA : public AudioDecoder {
public:
AudioDecoderPcmA() {}
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -86,8 +92,11 @@
class AudioDecoderPcm16B : public AudioDecoder {
public:
AudioDecoderPcm16B();
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
@@ -112,8 +121,11 @@
public:
AudioDecoderIlbc();
virtual ~AudioDecoderIlbc();
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
virtual bool HasDecodePlc() const { return true; }
virtual int DecodePlc(int num_frames, int16_t* decoded);
virtual int Init();
@@ -129,8 +141,11 @@
public:
AudioDecoderG722();
virtual ~AudioDecoderG722();
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ 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;
@@ -144,8 +159,11 @@
public:
AudioDecoderG722Stereo();
virtual ~AudioDecoderG722Stereo();
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type);
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type);
virtual int Init();
private:
@@ -169,10 +187,16 @@
public:
explicit AudioDecoderOpus(int num_channels);
virtual ~AudioDecoderOpus();
- 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 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 Init();
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
virtual int PacketDurationRedundant(const uint8_t* encoded,
@@ -195,8 +219,13 @@
public:
explicit AudioDecoderCng();
virtual ~AudioDecoderCng();
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) { return -1; }
+ 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 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 95805d3..1f0e881 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -186,10 +186,9 @@
// 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,
- &decoded[processed_samples * channels_],
- &speech_type);
+ size_t dec_len = decoder_->Decode(
+ &encoded_[encoded_bytes_], enc_len, codec_input_rate_hz_,
+ &decoded[processed_samples * channels_], &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
encoded_bytes_ += enc_len;
processed_samples += frame_size_;
@@ -222,13 +221,15 @@
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, output1.get(), &speech_type1);
+ dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
+ 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, output2.get(), &speech_type2);
+ dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
+ 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) {
@@ -247,8 +248,8 @@
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, output.get(), &speech_type);
+ size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
+ 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
@@ -338,8 +339,8 @@
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, output.get(), &speech_type);
+ size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
+ 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 503e46f..7288f11 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -22,8 +22,9 @@
MockAudioDecoder() {}
virtual ~MockAudioDecoder() { Die(); }
MOCK_METHOD0(Die, void());
- MOCK_METHOD4(Decode, int(const uint8_t*, size_t, int16_t*,
- AudioDecoder::SpeechType*));
+ MOCK_METHOD5(
+ Decode,
+ int(const uint8_t*, size_t, int, 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 19f069a..22d2816 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
@@ -29,8 +29,11 @@
public:
ExternalPcm16B() {}
- virtual int Decode(const uint8_t* encoded, size_t encoded_len,
- int16_t* decoded, SpeechType* speech_type) {
+ virtual int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) {
int16_t ret = WebRtcPcm16b_Decode(
encoded, static_cast<int16_t>(encoded_len), decoded);
*speech_type = ConvertSpeechType(1);
@@ -49,7 +52,7 @@
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));
@@ -65,9 +68,12 @@
virtual ~MockExternalPcm16B() { Die(); }
MOCK_METHOD0(Die, void());
- MOCK_METHOD4(Decode,
- int(const uint8_t* encoded, size_t encoded_len, int16_t* decoded,
- SpeechType* speech_type));
+ MOCK_METHOD5(Decode,
+ int(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ 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 a3dd271..0449044 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -100,7 +100,8 @@
next_arrival_time = GetArrivalTime(next_send_time);
} while (Lost()); // If lost, immediately read the next packet.
- EXPECT_CALL(*external_decoder_, Decode(_, payload_size_bytes_, _, _))
+ EXPECT_CALL(*external_decoder_,
+ Decode(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _))
.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 f1a3a90..7370825 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,
+ packet->payload, packet->payload_length, fs_hz_,
&decoded_buffer_[*decoded_length], speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
@@ -1274,10 +1274,9 @@
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
- decode_length = decoder->Decode(packet->payload,
- packet->payload_length,
- &decoded_buffer_[*decoded_length],
- speech_type);
+ decode_length =
+ decoder->Decode(packet->payload, packet->payload_length, fs_hz_,
+ &decoded_buffer_[*decoded_length], speech_type);
}
delete[] packet->payload;
@@ -1607,7 +1606,8 @@
if (decoder) {
const uint8_t* dummy_payload = NULL;
AudioDecoder::SpeechType speech_type;
- length = decoder->Decode(dummy_payload, 0, decoded_buffer, &speech_type);
+ length =
+ decoder->Decode(dummy_payload, 0, fs_hz_, 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 36ed35a..54b393b 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -430,6 +430,7 @@
// 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) {
@@ -521,10 +522,11 @@
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, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder,
+ Decode(Pointee(0), kPayloadLengthBytes, kSampleRateHz, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kSpeech),
+ SetArgPointee<4>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterExternalDecoder(
@@ -566,10 +568,11 @@
// 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, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder,
+ Decode(Pointee(2), kPayloadLengthBytes, kSampleRateHz, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kSpeech),
+ SetArgPointee<4>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
// Pull audio once.
@@ -682,28 +685,31 @@
// 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, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes,
+ kSampleRateKhz * 1000, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kSpeech),
+ SetArgPointee<4>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes,
+ kSampleRateKhz * 1000, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kComfortNoise),
+ SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, kSampleRateKhz * 1000, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kComfortNoise),
+ SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(kPayloadLengthSamples)));
- EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes, _, _))
- .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+ EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes,
+ kSampleRateKhz * 1000, _, _))
+ .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
- SetArgPointee<3>(AudioDecoder::kSpeech),
+ SetArgPointee<4>(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 b61bf83..cdcf0b3 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -36,16 +36,22 @@
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, int16_t* decoded,
+ int Decode(const uint8_t* encoded,
+ size_t encoded_len,
+ int /*sample_rate_hz*/,
+ 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,
- int16_t* decoded, SpeechType* speech_type) override {
- return Decode(encoded, encoded_len, decoded, speech_type);
+ 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 PacketDuration(const uint8_t* encoded,