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Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

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This should be safe to land now that issue 4143 was resolved (in r8492).
This change effectively reverts 8488.

TBR=kwiberg@webrtc.org

Original commit message:
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.

Review URL: https://webrtc-codereview.appspot.com/39289004

Cr-Commit-Position: refs/heads/master@{#8496}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-02-25 10:02:29 +00:00
parent 1e64263b90
commit 1eda4e3db6
20 changed files with 201 additions and 184 deletions
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21
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -186,10 +186,9 @@ class AudioDecoderTest : public ::testing::Test {
// 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 @@ class AudioDecoderTest : public ::testing::Test {
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 @@ class AudioDecoderTest : public ::testing::Test {
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 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
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()));