audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.h b/webrtc/modules/audio_coding/test/TestAllCodecs.h
new file mode 100644
index 0000000..e79bd69
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.h
@@ -0,0 +1,84 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Config;
+
+class TestPack : public AudioPacketizationCallback {
+ public:
+ TestPack();
+ ~TestPack();
+
+ void RegisterReceiverACM(AudioCodingModule* acm);
+
+ int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size,
+ const RTPFragmentationHeader* fragmentation) override;
+
+ size_t payload_size();
+ uint32_t timestamp_diff();
+ void reset_payload_size();
+
+ private:
+ AudioCodingModule* receiver_acm_;
+ uint16_t sequence_number_;
+ uint8_t payload_data_[60 * 32 * 2 * 2];
+ uint32_t timestamp_diff_;
+ uint32_t last_in_timestamp_;
+ uint64_t total_bytes_;
+ size_t payload_size_;
+};
+
+class TestAllCodecs : public ACMTest {
+ public:
+ explicit TestAllCodecs(int test_mode);
+ ~TestAllCodecs();
+
+ void Perform() override;
+
+ private:
+ // The default value of '-1' indicates that the registration is based only on
+ // codec name, and a sampling frequency matching is not required.
+ // This is useful for codecs which support several sampling frequency.
+ // Note! Only mono mode is tested in this test.
+ void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
+ int rate, int packet_size, size_t extra_byte);
+
+ void Run(TestPack* channel);
+ void OpenOutFile(int test_number);
+ void DisplaySendReceiveCodec();
+
+ int test_mode_;
+ rtc::scoped_ptr<AudioCodingModule> acm_a_;
+ rtc::scoped_ptr<AudioCodingModule> acm_b_;
+ TestPack* channel_a_to_b_;
+ PCMFile infile_a_;
+ PCMFile outfile_b_;
+ int test_count_;
+ int packet_size_samples_;
+ size_t packet_size_bytes_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_