Adding DTX logic to FakeDecodeFromFile (used be NetEqTest).

Bug: b/129521878
Change-Id: Ifcf868048a39ef1d2cc736988479f921e668167b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132799
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27626}
This commit is contained in:
Minyue Li
2019-04-15 14:13:08 +02:00
committed by Commit Bot
parent 72b7524d87
commit f9846bc172
2 changed files with 50 additions and 1 deletions

View File

@ -17,6 +17,53 @@
namespace webrtc {
namespace test {
namespace {
class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
public:
FakeEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
: decoder_(decoder), payload_(std::move(payload)) {}
size_t Duration() const override {
const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
return ret < 0 ? 0 : static_cast<size_t>(ret);
}
absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
auto speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
return ret < 0 ? absl::nullopt
: absl::optional<DecodeResult>(
{static_cast<size_t>(ret), speech_type});
}
// This is to mimic OpusFrame.
bool IsDtxPacket() const override {
uint32_t original_payload_size_bytes =
ByteReader<uint32_t>::ReadLittleEndian(&payload_.data()[8]);
return original_payload_size_bytes <= 2;
}
private:
AudioDecoder* const decoder_;
const rtc::Buffer payload_;
};
} // namespace
std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
std::unique_ptr<EncodedAudioFrame> frame(
new FakeEncodedFrame(this, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
return results;
}
int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,

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@ -20,7 +20,6 @@
namespace webrtc {
namespace test {
// Provides an AudioDecoder implementation that delivers audio data from a file.
// The "encoded" input should contain information about what RTP timestamp the
// encoding represents, and how many samples the decoder should produce for that
@ -38,6 +37,9 @@ class FakeDecodeFromFile : public AudioDecoder {
~FakeDecodeFromFile() = default;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp) override;
void Reset() override {}
int SampleRateHz() const override { return sample_rate_hz_; }