46d90dcd74
Adding support for 10, 40 and 60 ms packet sizes for Opus. BUG=issue1015 Review URL: https://webrtc-codereview.appspot.com/1086004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
118 lines
4.3 KiB
C++
118 lines
4.3 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
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#include <string.h>
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/typedefs.h"
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// Checks for enabled codecs, we prevent enabling codecs which are not
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// compatible.
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#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
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#error iSAC and iSACFX codecs cannot be enabled at the same time
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#endif
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#ifdef WIN32
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// OS-dependent case-insensitive string comparison
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#define STR_CASE_CMP(x, y) ::_stricmp(x, y)
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#else
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// OS-dependent case-insensitive string comparison
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#define STR_CASE_CMP(x, y) ::strcasecmp(x, y)
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#endif
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namespace webrtc {
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// 60 ms is the maximum block size we support. An extra 20 ms is considered
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// for safety if process() method is not called when it should be, i.e. we
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// accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
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#define AUDIO_BUFFER_SIZE_W16 7680
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// There is one timestamp per each 10 ms of audio
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// the audio buffer, at max, may contain 32 blocks of 10ms
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// audio if the sampling frequency is 8000 Hz (80 samples per block).
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// Therefore, The size of the buffer where we keep timestamps
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// is defined as follows
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#define TIMESTAMP_BUFFER_SIZE_W32 (AUDIO_BUFFER_SIZE_W16/80)
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// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
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#define MAX_PAYLOAD_SIZE_BYTE 7680
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// General codec specific defines
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const int kIsacWbDefaultRate = 32000;
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const int kIsacSwbDefaultRate = 56000;
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const int kIsacPacSize480 = 480;
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const int kIsacPacSize960 = 960;
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const int kIsacPacSize1440 = 1440;
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// An encoded bit-stream is labeled by one of the following enumerators.
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//
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// kNoEncoding : There has been no encoding.
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// kActiveNormalEncoded : Active audio frame coded by the codec.
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// kPassiveNormalEncoded : Passive audio frame coded by the codec.
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// kPassiveDTXNB : Passive audio frame coded by narrow-band CN.
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// kPassiveDTXWB : Passive audio frame coded by wide-band CN.
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// kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN.
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// kPassiveDTXFB : Passive audio frame coded by full-band CN.
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enum WebRtcACMEncodingType {
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kNoEncoding,
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kActiveNormalEncoded,
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kPassiveNormalEncoded,
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kPassiveDTXNB,
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kPassiveDTXWB,
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kPassiveDTXSWB,
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kPassiveDTXFB
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};
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// A structure which contains codec parameters. For instance, used when
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// initializing encoder and decoder.
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//
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// codec_inst: c.f. common_types.h
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// enable_dtx: set true to enable DTX. If codec does not have
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// internal DTX, this will enable VAD.
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// enable_vad: set true to enable VAD.
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// vad_mode: VAD mode, c.f. audio_coding_module_typedefs.h
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// for possible values.
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struct WebRtcACMCodecParams {
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CodecInst codec_inst;
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bool enable_dtx;
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bool enable_vad;
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ACMVADMode vad_mode;
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};
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// A structure that encapsulates audio buffer and related parameters
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// used for synchronization of audio of two ACMs.
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//
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// in_audio: same as ACMGenericCodec::in_audio_
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// in_audio_ix_read: same as ACMGenericCodec::in_audio_ix_read_
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// in_audio_ix_write: same as ACMGenericCodec::in_audio_ix_write_
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// in_timestamp: same as ACMGenericCodec::in_timestamp_
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// in_timestamp_ix_write: same as ACMGenericCodec::in_timestamp_ix_write_
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// last_timestamp: same as ACMGenericCodec::last_timestamp_
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// last_in_timestamp: same as AudioCodingModuleImpl::last_in_timestamp_
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//
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struct WebRtcACMAudioBuff {
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WebRtc_Word16 in_audio[AUDIO_BUFFER_SIZE_W16];
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WebRtc_Word16 in_audio_ix_read;
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WebRtc_Word16 in_audio_ix_write;
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WebRtc_UWord32 in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
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WebRtc_Word16 in_timestamp_ix_write;
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WebRtc_UWord32 last_timestamp;
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WebRtc_UWord32 last_in_timestamp;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
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