Revert r4772 "Compile ACM1 and ACM2."
Breaks Android build. TBR=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2244004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4777 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_amr.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_celt.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -12,7 +12,7 @@
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#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -17,7 +17,7 @@
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// references, where appropriate.
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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// Includes needed to create the codecs.
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113
webrtc/modules/audio_coding/main/source/acm_common_defs.h
Normal file
113
webrtc/modules/audio_coding/main/source/acm_common_defs.h
Normal file
@ -0,0 +1,113 @@
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/*
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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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namespace webrtc {
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namespace acm1 {
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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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int16_t in_audio[AUDIO_BUFFER_SIZE_W16];
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int16_t in_audio_ix_read;
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int16_t in_audio_ix_write;
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uint32_t in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
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int16_t in_timestamp_ix_write;
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uint32_t last_timestamp;
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uint32_t last_in_timestamp;
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};
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} // namespace acm1
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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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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -12,7 +12,7 @@
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#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_g7221.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_g729.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_g7291.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -16,7 +16,7 @@
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#include "webrtc/common_audio/vad/include/webrtc_vad.h"
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#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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@ -12,7 +12,7 @@
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -9,7 +9,7 @@
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*/
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#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_isac.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_pcma.h"
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h"
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -10,7 +10,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_red.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
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@ -11,7 +11,7 @@
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#include "webrtc/modules/audio_coding/main/source/acm_speex.h"
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#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
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||||
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
|
||||
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
|
||||
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
|
||||
|
||||
112
webrtc/modules/audio_coding/main/source/audio_coding_module.cc
Normal file
112
webrtc/modules/audio_coding/main/source/audio_coding_module.cc
Normal file
@ -0,0 +1,112 @@
|
||||
/*
|
||||
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
|
||||
|
||||
#include "webrtc/common_types.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
|
||||
#include "webrtc/system_wrappers/interface/clock.h"
|
||||
#include "webrtc/system_wrappers/interface/trace.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
// Create module
|
||||
AudioCodingModule* AudioCodingModule::Create(const int32_t id) {
|
||||
return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
|
||||
}
|
||||
|
||||
// Used for testing by inserting a simulated clock. ACM will not destroy the
|
||||
// injected |clock| the client has to take care of that.
|
||||
AudioCodingModule* AudioCodingModule::Create(const int32_t id,
|
||||
Clock* clock) {
|
||||
return new acm1::AudioCodingModuleImpl(id, clock);
|
||||
}
|
||||
|
||||
// Destroy module
|
||||
void AudioCodingModule::Destroy(AudioCodingModule* module) {
|
||||
delete static_cast<acm1::AudioCodingModuleImpl*>(module);
|
||||
}
|
||||
|
||||
// Get number of supported codecs
|
||||
uint8_t AudioCodingModule::NumberOfCodecs() {
|
||||
return static_cast<uint8_t>(acm1::ACMCodecDB::kNumCodecs);
|
||||
}
|
||||
|
||||
// Get supported codec param with id
|
||||
int32_t AudioCodingModule::Codec(uint8_t list_id,
|
||||
CodecInst* codec) {
|
||||
// Get the codec settings for the codec with the given list ID
|
||||
return acm1::ACMCodecDB::Codec(list_id, codec);
|
||||
}
|
||||
|
||||
// Get supported codec Param with name, frequency and number of channels.
|
||||
int32_t AudioCodingModule::Codec(const char* payload_name,
|
||||
CodecInst* codec, int sampling_freq_hz,
|
||||
int channels) {
|
||||
int codec_id;
|
||||
|
||||
// Get the id of the codec from the database.
|
||||
codec_id = acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz,
|
||||
channels);
|
||||
if (codec_id < 0) {
|
||||
// We couldn't find a matching codec, set the parameters to unacceptable
|
||||
// values and return.
|
||||
codec->plname[0] = '\0';
|
||||
codec->pltype = -1;
|
||||
codec->pacsize = 0;
|
||||
codec->rate = 0;
|
||||
codec->plfreq = 0;
|
||||
return -1;
|
||||
}
|
||||
|
||||
// Get default codec settings.
|
||||
acm1::ACMCodecDB::Codec(codec_id, codec);
|
||||
|
||||
// Keep the number of channels from the function call. For most codecs it
|
||||
// will be the same value as in default codec settings, but not for all.
|
||||
codec->channels = channels;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
// Get supported codec Index with name, frequency and number of channels.
|
||||
int32_t AudioCodingModule::Codec(const char* payload_name,
|
||||
int sampling_freq_hz, int channels) {
|
||||
return acm1::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
|
||||
}
|
||||
|
||||
// Checks the validity of the parameters of the given codec
|
||||
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
|
||||
int mirror_id;
|
||||
|
||||
int codec_number = acm1::ACMCodecDB::CodecNumber(&codec, &mirror_id);
|
||||
|
||||
if (codec_number < 0) {
|
||||
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
|
||||
"Invalid codec settings.");
|
||||
return false;
|
||||
} else {
|
||||
return true;
|
||||
}
|
||||
}
|
||||
|
||||
AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
|
||||
return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id),
|
||||
Clock::GetRealTimeClock());
|
||||
}
|
||||
|
||||
AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
|
||||
// TODO(minyue): return new AudioCodingModuleImpl (new version).
|
||||
return NULL;
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -37,7 +37,6 @@
|
||||
],
|
||||
'dependencies': [
|
||||
'<@(audio_coding_dependencies)',
|
||||
'acm2',
|
||||
],
|
||||
'include_dirs': [
|
||||
'../interface',
|
||||
@ -101,6 +100,7 @@
|
||||
'acm_red.h',
|
||||
'acm_resampler.cc',
|
||||
'acm_resampler.h',
|
||||
'audio_coding_module.cc',
|
||||
'audio_coding_module_impl.cc',
|
||||
'audio_coding_module_impl.h',
|
||||
'nack.cc',
|
||||
@ -146,7 +146,4 @@
|
||||
],
|
||||
}],
|
||||
],
|
||||
'includes': [
|
||||
'../acm2/audio_coding_module.gypi',
|
||||
],
|
||||
}
|
||||
|
||||
@ -17,7 +17,7 @@
|
||||
|
||||
#include "webrtc/engine_configurations.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
|
||||
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_dtmf_detection.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
|
||||
#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
|
||||
@ -1262,6 +1262,64 @@ int32_t AudioCodingModuleImpl::RegisterTransportCallback(
|
||||
return 0;
|
||||
}
|
||||
|
||||
// Used by the module to deliver messages to the codec module/application
|
||||
// AVT(DTMF).
|
||||
int32_t AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
|
||||
#ifndef WEBRTC_DTMF_DETECTION
|
||||
AudioCodingFeedback* /* incoming_message */,
|
||||
const ACMCountries /* cpt */) {
|
||||
return -1;
|
||||
#else
|
||||
AudioCodingFeedback* incoming_message,
|
||||
const ACMCountries cpt) {
|
||||
int16_t status = 0;
|
||||
|
||||
// Enter the critical section for callback.
|
||||
{
|
||||
CriticalSectionScoped lock(callback_crit_sect_);
|
||||
dtmf_callback_ = incoming_message;
|
||||
}
|
||||
// Enter the ACM critical section to set up the DTMF class.
|
||||
{
|
||||
CriticalSectionScoped lock(acm_crit_sect_);
|
||||
// Check if the call is to disable or enable the callback.
|
||||
if (incoming_message == NULL) {
|
||||
// Callback is disabled, delete DTMF-detector class.
|
||||
if (dtmf_detector_ != NULL) {
|
||||
delete dtmf_detector_;
|
||||
dtmf_detector_ = NULL;
|
||||
}
|
||||
status = 0;
|
||||
} else {
|
||||
status = 0;
|
||||
if (dtmf_detector_ == NULL) {
|
||||
dtmf_detector_ = new ACMDTMFDetection;
|
||||
if (dtmf_detector_ == NULL) {
|
||||
status = -1;
|
||||
}
|
||||
}
|
||||
if (status >= 0) {
|
||||
status = dtmf_detector_->Enable(cpt);
|
||||
if (status < 0) {
|
||||
// Failed to initialize if DTMF-detection was not enabled before,
|
||||
// delete the class, and set the callback to NULL and return -1.
|
||||
delete dtmf_detector_;
|
||||
dtmf_detector_ = NULL;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
// Check if we failed in setting up the DTMF-detector class.
|
||||
if ((status < 0)) {
|
||||
// We failed, we cannot have the callback.
|
||||
CriticalSectionScoped lock(callback_crit_sect_);
|
||||
dtmf_callback_ = NULL;
|
||||
}
|
||||
|
||||
return status;
|
||||
#endif
|
||||
}
|
||||
|
||||
// Add 10MS of raw (PCM) audio data to the encoder.
|
||||
int32_t AudioCodingModuleImpl::Add10MsData(
|
||||
const AudioFrame& audio_frame) {
|
||||
@ -2404,12 +2462,27 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
|
||||
return 0;
|
||||
}
|
||||
|
||||
/////////////////////////////////////////
|
||||
// (CNG) Comfort Noise Generation
|
||||
// Generate comfort noise when receiving DTX packets
|
||||
//
|
||||
|
||||
// Get VAD aggressiveness on the incoming stream
|
||||
ACMVADMode AudioCodingModuleImpl::ReceiveVADMode() const {
|
||||
return neteq_.vad_mode();
|
||||
}
|
||||
|
||||
// Configure VAD aggressiveness on the incoming stream.
|
||||
int16_t AudioCodingModuleImpl::SetReceiveVADMode(const ACMVADMode mode) {
|
||||
return neteq_.SetVADMode(mode);
|
||||
}
|
||||
|
||||
/////////////////////////////////////////
|
||||
// Statistics
|
||||
//
|
||||
|
||||
int32_t AudioCodingModuleImpl::NetworkStatistics(
|
||||
ACMNetworkStatistics* statistics) {
|
||||
ACMNetworkStatistics* statistics) const {
|
||||
int32_t status;
|
||||
status = neteq_.NetworkStatistics(statistics);
|
||||
return status;
|
||||
@ -2649,7 +2722,8 @@ int32_t AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
|
||||
return 0;
|
||||
}
|
||||
|
||||
int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
|
||||
int32_t AudioCodingModuleImpl::SetISACMaxRate(
|
||||
const uint32_t max_bit_per_sec) {
|
||||
CriticalSectionScoped lock(acm_crit_sect_);
|
||||
|
||||
if (!HaveValidEncoder("SetISACMaxRate")) {
|
||||
@ -2659,7 +2733,8 @@ int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
|
||||
return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec);
|
||||
}
|
||||
|
||||
int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
|
||||
int32_t AudioCodingModuleImpl::SetISACMaxPayloadSize(
|
||||
const uint16_t max_size_bytes) {
|
||||
CriticalSectionScoped lock(acm_crit_sect_);
|
||||
|
||||
if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
|
||||
@ -2671,9 +2746,9 @@ int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
|
||||
}
|
||||
|
||||
int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
|
||||
int frame_size_ms,
|
||||
int rate_bit_per_sec,
|
||||
bool enforce_frame_size) {
|
||||
const uint8_t frame_size_ms,
|
||||
const uint16_t rate_bit_per_sec,
|
||||
const bool enforce_frame_size) {
|
||||
CriticalSectionScoped lock(acm_crit_sect_);
|
||||
|
||||
if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
|
||||
@ -2684,6 +2759,21 @@ int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
|
||||
frame_size_ms, rate_bit_per_sec, enforce_frame_size);
|
||||
}
|
||||
|
||||
int32_t AudioCodingModuleImpl::SetBackgroundNoiseMode(
|
||||
const ACMBackgroundNoiseMode mode) {
|
||||
if ((mode < On) || (mode > Off)) {
|
||||
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
|
||||
"The specified background noise is out of range.\n");
|
||||
return -1;
|
||||
}
|
||||
return neteq_.SetBackgroundNoiseMode(mode);
|
||||
}
|
||||
|
||||
int32_t AudioCodingModuleImpl::BackgroundNoiseMode(
|
||||
ACMBackgroundNoiseMode* mode) {
|
||||
return neteq_.BackgroundNoiseMode(*mode);
|
||||
}
|
||||
|
||||
int32_t AudioCodingModuleImpl::PlayoutTimestamp(
|
||||
uint32_t* timestamp) {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
|
||||
@ -2719,7 +2809,8 @@ bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
|
||||
return true;
|
||||
}
|
||||
|
||||
int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
|
||||
int32_t AudioCodingModuleImpl::UnregisterReceiveCodec(
|
||||
const int16_t payload_type) {
|
||||
CriticalSectionScoped lock(acm_crit_sect_);
|
||||
int id;
|
||||
|
||||
|
||||
@ -23,14 +23,14 @@
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
struct WebRtcACMAudioBuff;
|
||||
struct WebRtcACMCodecParams;
|
||||
class CriticalSectionWrapper;
|
||||
class RWLockWrapper;
|
||||
class Clock;
|
||||
|
||||
namespace acm1 {
|
||||
|
||||
struct WebRtcACMAudioBuff;
|
||||
struct WebRtcACMCodecParams;
|
||||
class ACMDTMFDetection;
|
||||
class ACMGenericCodec;
|
||||
class Nack;
|
||||
@ -96,9 +96,20 @@ class AudioCodingModuleImpl : public AudioCodingModule {
|
||||
// called to deliver the encoded buffers.
|
||||
int32_t RegisterTransportCallback(AudioPacketizationCallback* transport);
|
||||
|
||||
// Used by the module to deliver messages to the codec module/application
|
||||
// AVT(DTMF).
|
||||
int32_t RegisterIncomingMessagesCallback(
|
||||
AudioCodingFeedback* incoming_message, const ACMCountries cpt);
|
||||
|
||||
// Add 10 ms of raw (PCM) audio data to the encoder.
|
||||
int32_t Add10MsData(const AudioFrame& audio_frame);
|
||||
|
||||
// Set background noise mode for NetEQ, on, off or fade.
|
||||
int32_t SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
|
||||
|
||||
// Get current background noise mode.
|
||||
int32_t BackgroundNoiseMode(ACMBackgroundNoiseMode* mode);
|
||||
|
||||
/////////////////////////////////////////
|
||||
// (FEC) Forward Error Correction
|
||||
//
|
||||
@ -123,6 +134,12 @@ class AudioCodingModuleImpl : public AudioCodingModule {
|
||||
|
||||
int32_t RegisterVADCallback(ACMVADCallback* vad_callback);
|
||||
|
||||
// Get VAD aggressiveness on the incoming stream.
|
||||
ACMVADMode ReceiveVADMode() const;
|
||||
|
||||
// Configure VAD aggressiveness on the incoming stream.
|
||||
int16_t SetReceiveVADMode(const ACMVADMode mode);
|
||||
|
||||
/////////////////////////////////////////
|
||||
// Receiver
|
||||
//
|
||||
@ -203,7 +220,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
|
||||
// Statistics
|
||||
//
|
||||
|
||||
int32_t NetworkStatistics(ACMNetworkStatistics* statistics);
|
||||
int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const;
|
||||
|
||||
void DestructEncoderInst(void* inst);
|
||||
|
||||
@ -226,16 +243,16 @@ class AudioCodingModuleImpl : public AudioCodingModule {
|
||||
|
||||
int32_t IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx);
|
||||
|
||||
int SetISACMaxRate(int max_bit_per_sec);
|
||||
int32_t SetISACMaxRate(const uint32_t max_bit_per_sec);
|
||||
|
||||
int SetISACMaxPayloadSize(int max_size_bytes);
|
||||
int32_t SetISACMaxPayloadSize(const uint16_t max_size_bytes);
|
||||
|
||||
int32_t ConfigISACBandwidthEstimator(
|
||||
int frame_size_ms,
|
||||
int rate_bit_per_sec,
|
||||
bool enforce_frame_size = false);
|
||||
const uint8_t frame_size_ms,
|
||||
const uint16_t rate_bit_per_sec,
|
||||
const bool enforce_frame_size = false);
|
||||
|
||||
int UnregisterReceiveCodec(uint8_t payload_type);
|
||||
int32_t UnregisterReceiveCodec(const int16_t payload_type);
|
||||
|
||||
std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
|
||||
|
||||
|
||||
Reference in New Issue
Block a user