Index: webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h |
deleted file mode 100644 |
index ab28fd96cf4b388d0cd33aba6da18812410d15e8..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h |
+++ /dev/null |
@@ -1,64 +0,0 @@ |
-/* |
- * 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. |
- */ |
- |
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ |
- |
-#include <memory> |
- |
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" |
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h" |
-#include "webrtc/modules/include/module_common_types.h" |
- |
-namespace webrtc { |
-namespace test { |
- |
-// This class provides a NetEqInput that takes audio from an input file and |
-// encodes it using a given audio encoder. |
-class EncodeNetEqInput : public NetEqInput { |
- public: |
- // The source will end after the given input duration. |
- EncodeNetEqInput(std::unique_ptr<InputAudioFile> input, |
- std::unique_ptr<AudioEncoder> encoder, |
- int64_t input_duration_ms); |
- |
- rtc::Optional<int64_t> NextPacketTime() const override; |
- |
- rtc::Optional<int64_t> NextOutputEventTime() const override; |
- |
- std::unique_ptr<PacketData> PopPacket() override; |
- |
- void AdvanceOutputEvent() override; |
- |
- bool ended() const override { |
- return next_output_event_ms_ <= input_duration_ms_; |
- } |
- |
- rtc::Optional<RTPHeader> NextHeader() const override; |
- |
- private: |
- static constexpr int64_t kOutputPeriodMs = 10; |
- |
- void CreatePacket(); |
- |
- std::unique_ptr<InputAudioFile> input_; |
- std::unique_ptr<AudioEncoder> encoder_; |
- std::unique_ptr<PacketData> packet_data_; |
- int32_t rtp_timestamp_ = 0; |
- int16_t sequence_number_ = 0; |
- int64_t next_packet_time_ms_ = 0; |
- int64_t next_output_event_ms_ = 0; |
- const int64_t input_duration_ms_; |
-}; |
- |
-} // namespace test |
-} // namespace webrtc |
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ |