Index: webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h |
deleted file mode 100644 |
index 3e65ec6c2d9c29f177d100e7beb53ba9c0c33a2c..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h |
+++ /dev/null |
@@ -1,91 +0,0 @@ |
-/* |
- * Copyright (c) 2014 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_MAIN_ACM2_ACM_SEND_TEST_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ |
- |
-#include <vector> |
- |
-#include "webrtc/base/constructormagic.h" |
-#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" |
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
-#include "webrtc/system_wrappers/include/clock.h" |
- |
-namespace webrtc { |
-class AudioEncoder; |
- |
-namespace test { |
-class InputAudioFile; |
-class Packet; |
- |
-class AcmSendTestOldApi : public AudioPacketizationCallback, |
- public PacketSource { |
- public: |
- AcmSendTestOldApi(InputAudioFile* audio_source, |
- int source_rate_hz, |
- int test_duration_ms); |
- virtual ~AcmSendTestOldApi() {} |
- |
- // Registers the send codec. Returns true on success, false otherwise. |
- bool RegisterCodec(const char* payload_name, |
- int sampling_freq_hz, |
- int channels, |
- int payload_type, |
- int frame_size_samples); |
- |
- // Registers an external send codec. Returns true on success, false otherwise. |
- bool RegisterExternalCodec(AudioEncoder* external_speech_encoder); |
- |
- // Returns the next encoded packet. Returns NULL if the test duration was |
- // exceeded. Ownership of the packet is handed over to the caller. |
- // Inherited from PacketSource. |
- Packet* NextPacket(); |
- |
- // Inherited from AudioPacketizationCallback. |
- int32_t SendData(FrameType frame_type, |
- uint8_t payload_type, |
- uint32_t timestamp, |
- const uint8_t* payload_data, |
- size_t payload_len_bytes, |
- const RTPFragmentationHeader* fragmentation) override; |
- |
- AudioCodingModule* acm() { return acm_.get(); } |
- |
- private: |
- static const int kBlockSizeMs = 10; |
- |
- // Creates a Packet object from the last packet produced by ACM (and received |
- // through the SendData method as a callback). Ownership of the new Packet |
- // object is transferred to the caller. |
- Packet* CreatePacket(); |
- |
- SimulatedClock clock_; |
- rtc::scoped_ptr<AudioCodingModule> acm_; |
- InputAudioFile* audio_source_; |
- int source_rate_hz_; |
- const size_t input_block_size_samples_; |
- AudioFrame input_frame_; |
- bool codec_registered_; |
- int test_duration_ms_; |
- // The following member variables are set whenever SendData() is called. |
- FrameType frame_type_; |
- int payload_type_; |
- uint32_t timestamp_; |
- uint16_t sequence_number_; |
- std::vector<uint8_t> last_payload_vec_; |
- bool data_to_send_; |
- |
- RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi); |
-}; |
- |
-} // namespace test |
-} // namespace webrtc |
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ |