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}diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index a51dac3..aad6b85 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -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,
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index c266d38..0260981 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -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 @@
~FakeDecodeFromFile() = default;
+ std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp) override;
+
void Reset() override {}
int SampleRateHz() const override { return sample_rate_hz_; }
