Get rid of unused types and constants in acm_common_defs.h

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1311743003 .

Cr-Commit-Position: refs/heads/master@{#9779}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
index 85a287e..208a50c 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
@@ -11,12 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
 
-#include <string.h>
-
-#include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/typedefs.h"
 
 // Checks for enabled codecs, we prevent enabling codecs which are not
 // compatible.
@@ -24,23 +19,10 @@
 #error iSAC and iSACFX codecs cannot be enabled at the same time
 #endif
 
-
 namespace webrtc {
 
-// 60 ms is the maximum block size we support. An extra 20 ms is considered
-// for safety if process() method is not called when it should be, i.e. we
-// accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
-#define AUDIO_BUFFER_SIZE_W16 7680
-
-// There is one timestamp per each 10 ms of audio
-// the audio buffer, at max, may contain 32 blocks of 10ms
-// audio if the sampling frequency is 8000 Hz (80 samples per block).
-// Therefore, The size of the buffer where we keep timestamps
-// is defined as follows
-#define TIMESTAMP_BUFFER_SIZE_W32  (AUDIO_BUFFER_SIZE_W16/80)
-
 // The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
-#define MAX_PAYLOAD_SIZE_BYTE   7680
+#define MAX_PAYLOAD_SIZE_BYTE 7680
 
 // General codec specific defines
 const int kIsacWbDefaultRate = 32000;
@@ -49,33 +31,6 @@
 const int kIsacPacSize960 = 960;
 const int kIsacPacSize1440 = 1440;
 
-// A structure which contains codec parameters. For instance, used when
-// initializing encoder and decoder.
-//
-//   codec_inst: c.f. common_types.h
-//   enable_dtx: set true to enable DTX. If codec does not have
-//               internal DTX, this will enable VAD.
-//   enable_vad: set true to enable VAD.
-//   vad_mode: VAD mode, c.f. audio_coding_module_typedefs.h
-//             for possible values.
-struct WebRtcACMCodecParams {
-  CodecInst codec_inst;
-  bool enable_dtx;
-  bool enable_vad;
-  ACMVADMode vad_mode;
-};
-
-// TODO(turajs): Remove when ACM1 is removed.
-struct WebRtcACMAudioBuff {
-  int16_t in_audio[AUDIO_BUFFER_SIZE_W16];
-  int16_t in_audio_ix_read;
-  int16_t in_audio_ix_write;
-  uint32_t in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
-  int16_t in_timestamp_ix_write;
-  uint32_t last_timestamp;
-  uint32_t last_in_timestamp;
-};
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_