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Get rid of unused types and constants in acm_common_defs.h

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R=henrik.lundin@webrtc.org

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

Cr-Commit-Position: refs/heads/master@{#9779}
This commit is contained in:
Karl Wiberg
2015-08-25 17:31:48 +02:00
parent 1bb8cf846d
commit f4772ee436
1 changed files with 1 additions and 46 deletions
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47
webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
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@ -11,12 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
#define 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/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 // Checks for enabled codecs, we prevent enabling codecs which are not
// compatible. // compatible.
@ -24,23 +19,10 @@
#error iSAC and iSACFX codecs cannot be enabled at the same time #error iSAC and iSACFX codecs cannot be enabled at the same time
#endif #endif
namespace webrtc { 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 // 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 // General codec specific defines
const int kIsacWbDefaultRate = 32000; const int kIsacWbDefaultRate = 32000;
@ -49,33 +31,6 @@ const int kIsacPacSize480 = 480;
const int kIsacPacSize960 = 960; const int kIsacPacSize960 = 960;
const int kIsacPacSize1440 = 1440; 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 } // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_