Index: webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..9ca5fb810bd64f21298680634793cc1b53a55ae5 |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc |
@@ -0,0 +1,63 @@ |
+/* |
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
+ |
+#include "webrtc/base/checks.h" |
+#include "webrtc/base/safe_conversions.h" |
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, |
+ size_t encoded_len, |
+ int /*sample_rate_hz*/, |
+ int16_t* decoded, |
+ SpeechType* speech_type) { |
+ RTC_CHECK_GE(encoded_len, 8u); |
+ uint32_t timestamp_to_decode = |
+ ByteReader<uint32_t>::ReadLittleEndian(encoded); |
+ uint32_t samples_to_decode = |
+ ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]); |
+ |
+ if (next_timestamp_from_input_ && |
+ timestamp_to_decode != *next_timestamp_from_input_) { |
+ // A gap in the timestamp sequence is detected. Skip the same number of |
+ // samples from the file. |
+ uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_; |
+ RTC_CHECK(input_->Seek(jump)); |
+ } |
+ |
+ RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded)); |
+ next_timestamp_from_input_ = |
+ rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode); |
+ |
+ if (stereo_) { |
+ InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2, |
+ decoded); |
+ samples_to_decode *= 2; |
+ } |
+ |
+ *speech_type = kSpeech; |
+ return samples_to_decode; |
+} |
+ |
+void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp, |
+ size_t samples, |
+ rtc::ArrayView<uint8_t> encoded) { |
+ RTC_CHECK_GE(encoded.size(), 8u); |
+ ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp); |
+ ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4], |
+ rtc::checked_cast<uint32_t>(samples)); |
+} |
+ |
+} // namespace test |
+} // namespace webrtc |