Index: webrtc/modules/audio_coding/main/acm2/acm_common_defs.h |
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 85a287e1268424056e053af13b6b0447d31ef2fa..208a50c7c2fd4f2e4791fdd21a12734b2e1d3ad8 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 kIsacPacSize480 = 480; |
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_ |