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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
commit43a97878ce14b72f0981164f87f2e35e14151312 (patch)
tree620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h
parentInitial commit. (diff)
downloadfirefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz
firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h')
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diff --git a/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h b/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h
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+/*
+ * Copyright (c) 2012 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 MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/RTPFile.h"
+#include "modules/include/module_common_types.h"
+
+namespace webrtc {
+
+#define MAX_INCOMING_PAYLOAD 8096
+
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization : public AudioPacketizationCallback {
+ public:
+ TestPacketization(RTPStream* rtpStream, uint16_t frequency);
+ ~TestPacketization();
+ int32_t SendData(AudioFrameType frameType,
+ uint8_t payloadType,
+ uint32_t timeStamp,
+ const uint8_t* payloadData,
+ size_t payloadSize,
+ int64_t absolute_capture_timestamp_ms) override;
+
+ private:
+ static void MakeRTPheader(uint8_t* rtpHeader,
+ uint8_t payloadType,
+ int16_t seqNo,
+ uint32_t timeStamp,
+ uint32_t ssrc);
+ RTPStream* _rtpStream;
+ int32_t _frequency;
+ int16_t _seqNo;
+};
+
+class Sender {
+ public:
+ Sender();
+ void Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ absl::string_view in_file_name,
+ int in_sample_rate,
+ int payload_type,
+ SdpAudioFormat format);
+ void Teardown();
+ void Run();
+ bool Add10MsData();
+
+ protected:
+ AudioCodingModule* _acm;
+
+ private:
+ PCMFile _pcmFile;
+ AudioFrame _audioFrame;
+ TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+ Receiver();
+ virtual ~Receiver() {}
+ void Setup(AudioCodingModule* acm,
+ RTPStream* rtpStream,
+ absl::string_view out_file_name,
+ size_t channels,
+ int file_num);
+ void Teardown();
+ void Run();
+ virtual bool IncomingPacket();
+ bool PlayoutData();
+
+ private:
+ PCMFile _pcmFile;
+ int16_t* _playoutBuffer;
+ uint16_t _playoutLengthSmpls;
+ int32_t _frequency;
+ bool _firstTime;
+
+ protected:
+ AudioCodingModule* _acm;
+ uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+ RTPStream* _rtpStream;
+ RTPHeader _rtpHeader;
+ size_t _realPayloadSizeBytes;
+ size_t _payloadSizeBytes;
+ uint32_t _nextTime;
+};
+
+class EncodeDecodeTest {
+ public:
+ EncodeDecodeTest();
+ void Perform();
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_