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87e2d785a0
This cl deprecates the FrameType enum, and adds aliases AudioFrameType and VideoFrameType. After downstream usage is updated, the enums will be separated and be moved out of common_types.h. Bug: webrtc:6883 Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27011}
90 lines
2.9 KiB
C++
90 lines
2.9 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_frame.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "rtc_base/constructor_magic.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class AudioEncoder;
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namespace test {
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class InputAudioFile;
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class Packet;
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class AcmSendTestOldApi : public AudioPacketizationCallback,
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public PacketSource {
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public:
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AcmSendTestOldApi(InputAudioFile* audio_source,
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int source_rate_hz,
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int test_duration_ms);
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~AcmSendTestOldApi() override;
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// Registers the send codec. Returns true on success, false otherwise.
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bool RegisterCodec(const char* payload_name,
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int sampling_freq_hz,
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int channels,
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int payload_type,
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int frame_size_samples);
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// Registers an external send codec.
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void RegisterExternalCodec(
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std::unique_ptr<AudioEncoder> external_speech_encoder);
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// Inherited from PacketSource.
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std::unique_ptr<Packet> NextPacket() override;
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// Inherited from AudioPacketizationCallback.
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int32_t SendData(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_len_bytes,
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const RTPFragmentationHeader* fragmentation) override;
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AudioCodingModule* acm() { return acm_.get(); }
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private:
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static const int kBlockSizeMs = 10;
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// Creates a Packet object from the last packet produced by ACM (and received
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// through the SendData method as a callback).
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std::unique_ptr<Packet> CreatePacket();
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SimulatedClock clock_;
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std::unique_ptr<AudioCodingModule> acm_;
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InputAudioFile* audio_source_;
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int source_rate_hz_;
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const size_t input_block_size_samples_;
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AudioFrame input_frame_;
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bool codec_registered_;
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int test_duration_ms_;
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// The following member variables are set whenever SendData() is called.
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AudioFrameType frame_type_;
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int payload_type_;
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uint32_t timestamp_;
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uint16_t sequence_number_;
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std::vector<uint8_t> last_payload_vec_;
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bool data_to_send_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
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