Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851
Original change's description:
> Remove CodecInst pt.1
>
> Update audio_coding tests to not use CodecInst.
>
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}
Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 9132d71..df6ee5f 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -47,13 +47,12 @@
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int sample_rate, size_t channels);
+ std::string in_file_name, int in_sample_rate,
+ int payload_type, SdpAudioFormat format);
void Teardown();
void Run();
bool Add10MsData();
- uint8_t codeId;
-
protected:
AudioCodingModule* _acm;
@@ -68,15 +67,12 @@
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels);
+ std::string out_file_name, size_t channels, int file_num);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
- //for auto_test and logging
- uint8_t codeId;
-
private:
PCMFile _pcmFile;
int16_t* _playoutBuffer;
@@ -96,17 +92,8 @@
class EncodeDecodeTest {
public:
- explicit EncodeDecodeTest(int test_mode);
+ EncodeDecodeTest();
void Perform();
-
- uint16_t _playoutFreq;
-
- private:
- std::string EncodeToFile(int fileType, int codeId, int* codePars);
-
- protected:
- Sender _sender;
- Receiver _receiver;
};
} // namespace webrtc