Compile ACM2 and ACM1.
First patch set is the same as patch set 3 of http://review.webrtc.org/2237004/
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these
APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these
APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF
detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded.
Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4785 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
index 8e14fbb..f99c85b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
@@ -102,26 +102,9 @@
namespace webrtc {
-// We dynamically allocate some of the dynamic payload types to the defined
-// codecs. Note! There are a limited number of payload types. If more codecs
-// are defined they will receive reserved fixed payload types (values 69-95).
-const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
- 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 92,
- 91, 90, 89, 88, 87, 86, 85, 84, 83, 82, 81, 80,
- 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
- 67, 66, 65
-};
-
-// Creates database with all supported codecs at compile time.
-// Each entry needs the following parameters in the given order:
-// payload type, name, sampling frequency, packet size in samples,
-// number of channels, and default rate.
-#if (defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) || \
- defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G722_1) || \
- defined(WEBRTC_CODEC_G722_1C) || defined(WEBRTC_CODEC_G729_1) || \
- defined(WEBRTC_CODEC_PCM16) || defined(WEBRTC_CODEC_SPEEX))
-static int count_database = 0;
-#endif
+// Not yet used payload-types.
+// 83, 82, 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
+// 67, 66, 65
const CodecInst ACMCodecDB::database_[] = {
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
@@ -133,13 +116,13 @@
#endif
#ifdef WEBRTC_CODEC_PCM16
// Mono
- {kDynamicPayloadtypes[count_database++], "L16", 8000, 80, 1, 128000},
- {kDynamicPayloadtypes[count_database++], "L16", 16000, 160, 1, 256000},
- {kDynamicPayloadtypes[count_database++], "L16", 32000, 320, 1, 512000},
+ {107, "L16", 8000, 80, 1, 128000},
+ {108, "L16", 16000, 160, 1, 256000},
+ {109, "L16", 32000, 320, 1, 512000},
// Stereo
- {kDynamicPayloadtypes[count_database++], "L16", 8000, 80, 2, 128000},
- {kDynamicPayloadtypes[count_database++], "L16", 16000, 160, 2, 256000},
- {kDynamicPayloadtypes[count_database++], "L16", 32000, 320, 2, 512000},
+ {111, "L16", 8000, 80, 2, 128000},
+ {112, "L16", 16000, 160, 2, 256000},
+ {113, "L16", 32000, 320, 2, 512000},
#endif
// G.711, PCM mu-law and A-law.
// Mono
@@ -152,16 +135,16 @@
{102, "ILBC", 8000, 240, 1, 13300},
#endif
#ifdef WEBRTC_CODEC_AMR
- {kDynamicPayloadtypes[count_database++], "AMR", 8000, 160, 1, 12200},
+ {114, "AMR", 8000, 160, 1, 12200},
#endif
#ifdef WEBRTC_CODEC_AMRWB
- {kDynamicPayloadtypes[count_database++], "AMR-WB", 16000, 320, 1, 20000},
+ {115, "AMR-WB", 16000, 320, 1, 20000},
#endif
#ifdef WEBRTC_CODEC_CELT
// Mono
- {kDynamicPayloadtypes[count_database++], "CELT", 32000, 640, 1, 64000},
+ {116, "CELT", 32000, 640, 1, 64000},
// Stereo
- {kDynamicPayloadtypes[count_database++], "CELT", 32000, 640, 2, 64000},
+ {117, "CELT", 32000, 640, 2, 64000},
#endif
#ifdef WEBRTC_CODEC_G722
// Mono
@@ -170,20 +153,20 @@
{119, "G722", 16000, 320, 2, 64000},
#endif
#ifdef WEBRTC_CODEC_G722_1
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 32000},
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 24000},
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 16000},
+ {92, "G7221", 16000, 320, 1, 32000},
+ {91, "G7221", 16000, 320, 1, 24000},
+ {90, "G7221", 16000, 320, 1, 16000},
#endif
#ifdef WEBRTC_CODEC_G722_1C
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 48000},
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 32000},
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 24000},
+ {89, "G7221", 32000, 640, 1, 48000},
+ {88, "G7221", 32000, 640, 1, 32000},
+ {87, "G7221", 32000, 640, 1, 24000},
#endif
#ifdef WEBRTC_CODEC_G729
{18, "G729", 8000, 240, 1, 8000},
#endif
#ifdef WEBRTC_CODEC_G729_1
- {kDynamicPayloadtypes[count_database++], "G7291", 16000, 320, 1, 32000},
+ {86, "G7291", 16000, 320, 1, 32000},
#endif
#ifdef WEBRTC_CODEC_GSMFR
{3, "GSM", 8000, 160, 1, 13200},
@@ -194,14 +177,16 @@
{120, "opus", 48000, 960, 2, 64000},
#endif
#ifdef WEBRTC_CODEC_SPEEX
- {kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
- {kDynamicPayloadtypes[count_database++], "speex", 16000, 320, 1, 22000},
+ {85, "speex", 8000, 160, 1, 11000},
+ {84, "speex", 16000, 320, 1, 22000},
#endif
// Comfort noise for four different sampling frequencies.
{13, "CN", 8000, 240, 1, 0},
{98, "CN", 16000, 480, 1, 0},
{99, "CN", 32000, 960, 1, 0},
+#ifdef ENABLE_48000_HZ
{100, "CN", 48000, 1440, 1, 0},
+#endif
#ifdef WEBRTC_CODEC_AVT
{106, "telephone-event", 8000, 240, 1, 0},
#endif
@@ -295,7 +280,9 @@
{1, {240}, 240, 1, false},
{1, {480}, 480, 1, false},
{1, {960}, 960, 1, false},
+#ifdef ENABLE_48000_HZ
{1, {1440}, 1440, 1, false},
+#endif
#ifdef WEBRTC_CODEC_AVT
{1, {240}, 240, 1, false},
#endif
@@ -383,8 +370,10 @@
// Comfort noise for three different sampling frequencies.
kDecoderCNGnb,
kDecoderCNGwb,
- kDecoderCNGswb32kHz,
- kDecoderCNGswb48kHz
+ kDecoderCNGswb32kHz
+#ifdef ENABLE_48000_HZ
+ , kDecoderCNGswb48kHz
+#endif
#ifdef WEBRTC_CODEC_AVT
, kDecoderAVT
#endif
@@ -710,10 +699,12 @@
codec_id = kCNSWB;
break;
}
+#ifdef ENABLE_48000_HZ
case 48000: {
codec_id = kCNFB;
break;
}
+#endif
default: {
return NULL;
}
@@ -765,10 +756,12 @@
codec_id = kCNSWB;
break;
}
+#ifdef ENABLE_48000_HZ
case 48000: {
codec_id = kCNFB;
break;
}
+#endif
default: {
return NULL;
}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
index a8a7643..b992b7d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
@@ -103,7 +103,9 @@
, kCNNB
, kCNWB
, kCNSWB
+#ifdef ENABLE_48000_HZ
, kCNFB
+#endif
#ifdef WEBRTC_CODEC_AVT
, kAVT
#endif
@@ -187,6 +189,9 @@
#ifndef WEBRTC_CODEC_RED
enum {kRED = -1};
#endif
+#ifndef ENABLE_48000_HZ
+ enum { kCNFB = -1 };
+#endif
// kMaxNumCodecs - Maximum number of codecs that can be activated in one
// build.
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 39287ea..26aa74d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
@@ -24,22 +24,13 @@
#error iSAC and iSACFX codecs cannot be enabled at the same time
#endif
-#ifndef STR_CASE_CMP
-#ifdef WIN32
-// OS-dependent case-insensitive string comparison
-#define STR_CASE_CMP(x, y) ::_stricmp(x, y)
-#else
-// OS-dependent case-insensitive string comparison
-#define STR_CASE_CMP(x, y) ::strcasecmp(x, y)
-#endif
-#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 @ 32 kHz (super wide-band) is 2560 samples.
-#define AUDIO_BUFFER_SIZE_W16 2560
+// 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
@@ -93,6 +84,17 @@
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_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index e2de7ef..7957fd3 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -301,7 +301,7 @@
uint32_t arrival_timestamp) {
return ACM_ISAC_DECODE_BWE(static_cast<ACM_ISAC_STRUCT*>(state_),
reinterpret_cast<const uint16_t*>(payload),
- payload_len,
+ static_cast<uint32_t>(payload_len),
rtp_sequence_number,
rtp_timestamp,
arrival_timestamp);
@@ -311,7 +311,7 @@
size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = ACM_ISAC_DECODERCU(static_cast<ISACStruct*>(state_),
+ int16_t ret = ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(state_),
reinterpret_cast<const uint16_t*>(encoded),
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 5a36f86..949a705 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -179,7 +179,7 @@
// improve performance. Here, this call has to be placed before the following
// block, therefore, we keep it inside critical section. Otherwise, we have to
// release |neteq_crit_sect_| and acquire it again, which seems an overkill.
- if (neteq_->SetMinimumDelay(delay_ms) < 0)
+ if (!neteq_->SetMinimumDelay(delay_ms))
return -1;
const int kLatePacketThreshold = 5;
@@ -593,7 +593,8 @@
int AcmReceiver::RedPayloadType() const {
CriticalSectionScoped lock(neteq_crit_sect_);
- if (!decoders_[ACMCodecDB::kRED].registered) {
+ if (ACMCodecDB::kRED < 0 ||
+ !decoders_[ACMCodecDB::kRED].registered) {
LOG_F(LS_WARNING) << "RED is not registered.";
return -1;
}
@@ -620,7 +621,7 @@
acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
- acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found;
+ acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
acm_stat->currentExpandRate = neteq_stat.expand_rate;
@@ -745,7 +746,7 @@
int max_num_packets;
int buffer_size_byte;
int max_buffer_size_byte;
- const float kBufferingThresholdScale = 0.9;
+ const float kBufferingThresholdScale = 0.9f;
neteq_->PacketBufferStatistics(&num_packets, &max_num_packets,
&buffer_size_byte, &max_buffer_size_byte);
if (num_packets > max_num_packets * kBufferingThresholdScale ||
@@ -786,7 +787,8 @@
int AcmReceiver::RtpHeaderToCodecIndex(
const RTPHeader &rtp_header, const uint8_t* payload) const {
uint8_t payload_type = rtp_header.payloadType;
- if (decoders_[ACMCodecDB::kRED].registered &&
+ if (ACMCodecDB::kRED >= 0 && // This ensures that RED is defined in WebRTC.
+ decoders_[ACMCodecDB::kRED].registered &&
payload_type == decoders_[ACMCodecDB::kRED].payload_type) {
// This is a RED packet, get the payload of the audio codec.
payload_type = payload[0] & 0x7F;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index 491160d..afb6920 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -13,18 +13,24 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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(int id) {
- return new AudioCodingModuleImpl(id);
+ return new acm1::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
+}
+
+AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
+ return new acm1::AudioCodingModuleImpl(id, clock);
}
// Destroy module
void AudioCodingModule::Destroy(AudioCodingModule* module) {
- delete static_cast<AudioCodingModuleImpl*>(module);
+ delete module;
}
// Get number of supported codecs
@@ -90,11 +96,12 @@
}
}
-AudioCodingModule* AudioCodingModuleFactory::Create(const int32_t id) const {
- return NULL;
+AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
+ return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id),
+ Clock::GetRealTimeClock());
}
-AudioCodingModule* NewAudioCodingModuleFactory::Create(const int32_t id) const {
+AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
return new AudioCodingModuleImpl(id);
}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
index f526250..e86fbfc 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -16,6 +16,7 @@
],
'dependencies': [
'<@(audio_coding_dependencies)',
+ 'NetEq4',
],
'include_dirs': [
'../interface',
@@ -40,6 +41,7 @@
'acm_cng.h',
'acm_codec_database.cc',
'acm_codec_database.h',
+ 'acm_common_defs.h',
'acm_dtmf_playout.cc',
'acm_dtmf_playout.h',
'acm_g722.cc',
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
index 038b132..ac79aa5 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
@@ -153,7 +153,6 @@
void InitialDelayManager::LatePackets(
uint32_t timestamp_now, SyncStream* sync_stream) {
assert(sync_stream);
- const int kLateThreshold = 5;
sync_stream->num_sync_packets = 0;
// If there is no estimate of timestamp increment, |timestamp_step_|, then
@@ -171,7 +170,7 @@
int num_late_packets = (timestamp_now - last_receive_timestamp_) /
timestamp_step_;
- if (num_late_packets < kLateThreshold)
+ if (num_late_packets < late_packet_threshold_)
return;
int sync_offset = 1; // One gap at the end of the sync-stream.
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 12b5a63..0074809 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -85,8 +85,8 @@
// injected into ACM. ACM will take the owner ship of the object clock and
// delete it when destroyed.
//
- static AudioCodingModule* Create(const int32_t id);
- static AudioCodingModule* Create(const int32_t id, Clock* clock);
+ static AudioCodingModule* Create(int id);
+ static AudioCodingModule* Create(int id, Clock* clock);
virtual ~AudioCodingModule() {};
// TODO(ajm): Deprecated. Remove all calls to this unneeded method.
@@ -103,7 +103,7 @@
// Return value:
// number of supported codecs.
///
- static uint8_t NumberOfCodecs();
+ static int NumberOfCodecs();
///////////////////////////////////////////////////////////////////////////
// int32_t Codec()
@@ -120,7 +120,7 @@
// -1 if the list number (list_id) is invalid.
// 0 if succeeded.
//
- static int32_t Codec(uint8_t list_id, CodecInst* codec);
+ static int Codec(int list_id, CodecInst* codec);
///////////////////////////////////////////////////////////////////////////
// int32_t Codec()
@@ -141,7 +141,7 @@
// -1 if no codec matches the given parameters.
// 0 if succeeded.
//
- static int32_t Codec(const char* payload_name, CodecInst* codec,
+ static int Codec(const char* payload_name, CodecInst* codec,
int sampling_freq_hz, int channels);
///////////////////////////////////////////////////////////////////////////
@@ -160,7 +160,7 @@
// if the codec is found, the index of the codec in the list,
// -1 if the codec is not found.
//
- static int32_t Codec(const char* payload_name, int sampling_freq_hz,
+ static int Codec(const char* payload_name, int sampling_freq_hz,
int channels);
///////////////////////////////////////////////////////////////////////////
@@ -582,8 +582,8 @@
// -1 if fails to unregister.
// 0 if the given codec is successfully unregistered.
//
- virtual int32_t UnregisterReceiveCodec(
- const int16_t payload_type) = 0;
+ virtual int UnregisterReceiveCodec(
+ uint8_t payload_type) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ReceiveCodec()
@@ -683,29 +683,6 @@
virtual int LeastRequiredDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
- // int32_t RegisterIncomingMessagesCallback()
- // Used by the module to deliver messages to the codec module/application
- // when a DTMF tone is detected, as well as when it stopped.
- //
- // Inputs:
- // -in_message_callback: pointer to callback function which will be called
- // if DTMF is detected.
- // -cpt : enables CPT (Call Progress Tone) detection for the
- // specified country. c.f. definition of ACMCountries
- // in audio_coding_module_typedefs.h for valid
- // entries. The default value disables CPT
- // detection.
- //
- // Return value:
- // -1 if the message callback could not be registered
- // 0 if registration is successful.
- //
- virtual int32_t
- RegisterIncomingMessagesCallback(
- AudioCodingFeedback* in_message_callback,
- const ACMCountries cpt = ACMDisableCountryDetection) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int32_t SetDtmfPlayoutStatus()
// Configure DTMF playout, i.e. whether out-of-band
// DTMF tones are played or not.
@@ -731,39 +708,6 @@
virtual bool DtmfPlayoutStatus() const = 0;
///////////////////////////////////////////////////////////////////////////
- // int32_t SetBackgroundNoiseMode()
- // Sets the mode of the background noise playout in an event of long
- // packet loss burst. For the valid modes see the declaration of
- // ACMBackgroundNoiseMode in audio_coding_module_typedefs.h.
- //
- // Input:
- // -mode : the mode for the background noise playout.
- //
- // Return value:
- // -1 if failed to set the mode.
- // 0 if succeeded in setting the mode.
- //
- virtual int32_t SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t BackgroundNoiseMode()
- // Call this method to get the mode of the background noise playout.
- // Playout of background noise is a result of a long packet loss burst.
- // See ACMBackgroundNoiseMode in audio_coding_module_typedefs.h for
- // possible modes.
- //
- // Output:
- // -mode : a reference to ACMBackgroundNoiseMode enumerator.
- //
- // Return value:
- // 0 if the output is a valid mode.
- // -1 if ACM failed to output a valid mode.
- //
- // TODO(tlegrand): Change function to return the mode.
- virtual int32_t BackgroundNoiseMode(ACMBackgroundNoiseMode* mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded
// audio of the packet in question. This function returns the timestamp of
@@ -853,39 +797,6 @@
AudioFrame* audio_frame) = 0;
///////////////////////////////////////////////////////////////////////////
- // (CNG) Comfort Noise Generation
- // Generate comfort noise when receiving DTX packets
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // int16_t SetReceiveVADMode()
- // Configure VAD aggressiveness on the incoming stream.
- //
- // Input:
- // -mode : aggressiveness of the VAD on incoming stream.
- // See audio_coding_module_typedefs.h for the
- // definition of ACMVADMode, and possible
- // values for aggressiveness.
- //
- // Return value:
- // -1 if fails to set the mode,
- // 0 if the mode is set successfully.
- //
- virtual int16_t SetReceiveVADMode(const ACMVADMode mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // ACMVADMode ReceiveVADMode()
- // Get VAD aggressiveness on the incoming stream.
- //
- // Return value:
- // aggressiveness of VAD, running on the incoming stream. A more
- // aggressive mode means more audio frames will be labeled as in-active.
- // See audio_coding_module_typedefs.h for the definition of
- // ACMVADMode.
- //
- virtual ACMVADMode ReceiveVADMode() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// Codec specific
//
@@ -904,8 +815,7 @@
// -1 if failed to set the maximum rate.
// 0 if the maximum rate is set successfully.
//
- virtual int32_t SetISACMaxRate(
- const uint32_t max_rate_bps) = 0;
+ virtual int SetISACMaxRate(int max_rate_bps) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SetISACMaxPayloadSize()
@@ -922,8 +832,7 @@
// -1 if failed to set the maximum payload-size.
// 0 if the given length is set successfully.
//
- virtual int32_t SetISACMaxPayloadSize(
- const uint16_t max_payload_len_bytes) = 0;
+ virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t ConfigISACBandwidthEstimator()
@@ -950,9 +859,9 @@
// 0 if the configuration was successfully applied.
//
virtual int32_t ConfigISACBandwidthEstimator(
- const uint8_t init_frame_size_ms,
- const uint16_t init_rate_bps,
- const bool enforce_frame_size = false) = 0;
+ int init_frame_size_ms,
+ int init_rate_bps,
+ bool enforce_frame_size = false) = 0;
///////////////////////////////////////////////////////////////////////////
// statistics
@@ -960,7 +869,8 @@
///////////////////////////////////////////////////////////////////////////
// int32_t NetworkStatistics()
- // Get network statistics.
+ // Get network statistics. Note that the internal statistics of NetEq are
+ // reset by this call.
//
// Input:
// -network_statistics : a structure that contains network statistics.
@@ -970,7 +880,7 @@
// 0 if statistics are set successfully.
//
virtual int32_t NetworkStatistics(
- ACMNetworkStatistics* network_statistics) const = 0;
+ ACMNetworkStatistics* network_statistics) = 0;
//
// Set an initial delay for playout.
diff --git a/webrtc/modules/audio_coding/main/source/acm_amr.cc b/webrtc/modules/audio_coding/main/source/acm_amr.cc
index 5590970..d398607 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amr.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_amr.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
index e2b7635..8b1b58d 100644
--- a/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_amrwb.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_amrwb.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_celt.cc b/webrtc/modules/audio_coding/main/source/acm_celt.cc
index 81a0346..3b83814 100644
--- a/webrtc/modules/audio_coding/main/source/acm_celt.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_celt.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_celt.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_cng.cc b/webrtc/modules/audio_coding/main/source/acm_cng.cc
index 57c48cd..6f3a505 100644
--- a/webrtc/modules/audio_coding/main/source/acm_cng.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_cng.cc
@@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
index c3a54d9..138effd 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
@@ -17,7 +17,7 @@
// references, where appropriate.
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/system_wrappers/interface/trace.h"
// Includes needed to create the codecs.
diff --git a/webrtc/modules/audio_coding/main/source/acm_common_defs.h b/webrtc/modules/audio_coding/main/source/acm_common_defs.h
deleted file mode 100644
index ecc41f8..0000000
--- a/webrtc/modules/audio_coding/main/source/acm_common_defs.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (c) 2011 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_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.
-#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
-#error iSAC and iSACFX codecs cannot be enabled at the same time
-#endif
-
-namespace webrtc {
-
-namespace acm1 {
-
-// 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
-
-// General codec specific defines
-const int kIsacWbDefaultRate = 32000;
-const int kIsacSwbDefaultRate = 56000;
-const int kIsacPacSize480 = 480;
-const int kIsacPacSize960 = 960;
-const int kIsacPacSize1440 = 1440;
-
-// An encoded bit-stream is labeled by one of the following enumerators.
-//
-// kNoEncoding : There has been no encoding.
-// kActiveNormalEncoded : Active audio frame coded by the codec.
-// kPassiveNormalEncoded : Passive audio frame coded by the codec.
-// kPassiveDTXNB : Passive audio frame coded by narrow-band CN.
-// kPassiveDTXWB : Passive audio frame coded by wide-band CN.
-// kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN.
-// kPassiveDTXFB : Passive audio frame coded by full-band CN.
-enum WebRtcACMEncodingType {
- kNoEncoding,
- kActiveNormalEncoded,
- kPassiveNormalEncoded,
- kPassiveDTXNB,
- kPassiveDTXWB,
- kPassiveDTXSWB,
- kPassiveDTXFB
-};
-
-// 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;
-};
-
-// A structure that encapsulates audio buffer and related parameters
-// used for synchronization of audio of two ACMs.
-//
-// in_audio: same as ACMGenericCodec::in_audio_
-// in_audio_ix_read: same as ACMGenericCodec::in_audio_ix_read_
-// in_audio_ix_write: same as ACMGenericCodec::in_audio_ix_write_
-// in_timestamp: same as ACMGenericCodec::in_timestamp_
-// in_timestamp_ix_write: same as ACMGenericCodec::in_timestamp_ix_write_
-// last_timestamp: same as ACMGenericCodec::last_timestamp_
-// last_in_timestamp: same as AudioCodingModuleImpl::last_in_timestamp_
-//
-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 acm1
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
index c8dea71..32195e6 100644
--- a/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_dtmf_playout.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_g722.cc b/webrtc/modules/audio_coding/main/source/acm_g722.cc
index 5368b35..1c19109 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g722.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g722.cc
@@ -12,7 +12,7 @@
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221.cc b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
index c9074ac..ed172fd 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_g7221.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
index 91071e9..96caba0 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7221c.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_g7221c.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_g729.cc b/webrtc/modules/audio_coding/main/source/acm_g729.cc
index 5b75ab9..406bb61 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g729.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g729.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_g729.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_g7291.cc b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
index fd241b3..0da6c99 100644
--- a/webrtc/modules/audio_coding/main/source/acm_g7291.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_g7291.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_g7291.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
index 52f5114..6c43301 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -16,7 +16,7 @@
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
#include "webrtc/system_wrappers/interface/trace.h"
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
index 3951a94..d6403f5 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
diff --git a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
index 9fa0410..5ea0c56 100644
--- a/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_gsmfr.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
index b47e750..0f8049e 100644
--- a/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_ilbc.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/modules/audio_coding/main/source/acm_ilbc.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_isac.cc b/webrtc/modules/audio_coding/main/source/acm_isac.cc
index b9316d6..61fa32f 100644
--- a/webrtc/modules/audio_coding/main/source/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_isac.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_isac.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc
index 3a619d0..413f371 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
index b0032b8..6fe12f7 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_pcm16b.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcma.cc b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
index c646417..9e5514a 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcma.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcma.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_pcma.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
index 5b6a457..6f4eb27 100644
--- a/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_pcmu.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_pcmu.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_red.cc b/webrtc/modules/audio_coding/main/source/acm_red.cc
index bc44c72..0d8134c 100644
--- a/webrtc/modules/audio_coding/main/source/acm_red.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_red.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_red.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/acm_speex.cc b/webrtc/modules/audio_coding/main/source/acm_speex.cc
index 5752693..1567929 100644
--- a/webrtc/modules/audio_coding/main/source/acm_speex.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_speex.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/audio_coding/main/source/acm_speex.h"
#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
deleted file mode 100644
index 9461a1f..0000000
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * 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
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
index 1709c17..94c3bcb 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -37,6 +37,7 @@
],
'dependencies': [
'<@(audio_coding_dependencies)',
+ 'acm2',
],
'include_dirs': [
'../interface',
@@ -100,7 +101,6 @@
'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',
@@ -146,4 +146,7 @@
],
}],
],
+ 'includes': [
+ '../acm2/audio_coding_module.gypi',
+ ],
}
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index 93b21e6..f5f8450 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -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/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
@@ -1262,64 +1262,6 @@
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) {
@@ -2463,26 +2405,11 @@
}
/////////////////////////////////////////
-// (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) const {
+ ACMNetworkStatistics* statistics) {
int32_t status;
status = neteq_.NetworkStatistics(statistics);
return status;
@@ -2722,8 +2649,7 @@
return 0;
}
-int32_t AudioCodingModuleImpl::SetISACMaxRate(
- const uint32_t max_bit_per_sec) {
+int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("SetISACMaxRate")) {
@@ -2733,8 +2659,7 @@
return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec);
}
-int32_t AudioCodingModuleImpl::SetISACMaxPayloadSize(
- const uint16_t max_size_bytes) {
+int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
@@ -2746,9 +2671,9 @@
}
int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
- const uint8_t frame_size_ms,
- const uint16_t rate_bit_per_sec,
- const bool enforce_frame_size) {
+ int frame_size_ms,
+ int rate_bit_per_sec,
+ bool enforce_frame_size) {
CriticalSectionScoped lock(acm_crit_sect_);
if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
@@ -2759,21 +2684,6 @@
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_,
@@ -2809,8 +2719,7 @@
return true;
}
-int32_t AudioCodingModuleImpl::UnregisterReceiveCodec(
- const int16_t payload_type) {
+int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
CriticalSectionScoped lock(acm_crit_sect_);
int id;
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
index 64afe4f..b63ae09 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -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;
@@ -96,20 +96,9 @@
// 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
//
@@ -134,12 +123,6 @@
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
//
@@ -220,7 +203,7 @@
// Statistics
//
- int32_t NetworkStatistics(ACMNetworkStatistics* statistics) const;
+ int32_t NetworkStatistics(ACMNetworkStatistics* statistics);
void DestructEncoderInst(void* inst);
@@ -243,16 +226,16 @@
int32_t IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx);
- int32_t SetISACMaxRate(const uint32_t max_bit_per_sec);
+ int SetISACMaxRate(int max_bit_per_sec);
- int32_t SetISACMaxPayloadSize(const uint16_t max_size_bytes);
+ int SetISACMaxPayloadSize(int max_size_bytes);
int32_t ConfigISACBandwidthEstimator(
- const uint8_t frame_size_ms,
- const uint16_t rate_bit_per_sec,
- const bool enforce_frame_size = false);
+ int frame_size_ms,
+ int rate_bit_per_sec,
+ bool enforce_frame_size = false);
- int32_t UnregisterReceiveCodec(const int16_t payload_type);
+ int UnregisterReceiveCodec(uint8_t payload_type);
std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/main/test/APITest.cc
index cb7115e..a9e2e71 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/webrtc/modules/audio_coding/main/test/APITest.cc
@@ -22,7 +22,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
@@ -98,11 +98,6 @@
_payloadUsed[n] = false;
}
- for (n = 0; n < 3; n++) {
- _receiveVADActivityA[n] = 0;
- _receiveVADActivityB[n] = 0;
- }
-
_movingDot[40] = '\0';
for (int n = 0; n < 40; n++) {
@@ -352,7 +347,6 @@
if (_writeToFile) {
_outFileA.Write10MsData(audioFrame);
}
- _receiveVADActivityA[(int) audioFrame.vad_activity_]++;
}
return true;
}
@@ -374,7 +368,6 @@
if (_writeToFile) {
_outFileB.Write10MsData(audioFrame);
}
- _receiveVADActivityB[(int) audioFrame.vad_activity_]++;
}
return true;
}
@@ -458,7 +451,7 @@
{
WriteLockScoped cs(_apiTestRWLock);
if (thread == 'A') {
- _testNumA = (_testNumB + 1 + (rand() % 6)) % 7;
+ _testNumA = (_testNumB + 1 + (rand() % 4)) % 5;
testNum = _testNumA;
_movingDot[_dotPositionA] = ' ';
@@ -471,7 +464,7 @@
_dotPositionA += _dotMoveDirectionA;
_movingDot[_dotPositionA] = (_dotMoveDirectionA > 0) ? '>' : '<';
} else {
- _testNumB = (_testNumA + 1 + (rand() % 6)) % 7;
+ _testNumB = (_testNumA + 1 + (rand() % 4)) % 5;
testNum = _testNumB;
_movingDot[_dotPositionB] = ' ';
@@ -507,14 +500,6 @@
case 4:
TestRegisteration('A');
break;
- case 5:
- TestReceiverVAD('A');
- break;
- case 6:
-#ifdef WEBRTC_DTMF_DETECTION
- LookForDTMF('A');
-#endif
- break;
default:
fprintf(stderr, "Wrong Test Number\n");
getchar();
@@ -543,10 +528,6 @@
// VAD TEST
TestSendVAD('A');
TestRegisteration('A');
- TestReceiverVAD('A');
-#ifdef WEBRTC_DTMF_DETECTION
- LookForDTMF('A');
-#endif
}
return true;
}
@@ -981,18 +962,15 @@
void APITest::TestPlayout(char receiveSide) {
AudioCodingModule* receiveACM;
AudioPlayoutMode* playoutMode = NULL;
- ACMBackgroundNoiseMode* bgnMode = NULL;
switch (receiveSide) {
case 'A': {
receiveACM = _acmA;
playoutMode = &_playoutModeA;
- bgnMode = &_bgnModeA;
break;
}
case 'B': {
receiveACM = _acmB;
playoutMode = &_playoutModeB;
- bgnMode = &_bgnModeB;
break;
}
default:
@@ -1005,29 +983,6 @@
CHECK_ERROR_MT(receiveFreqHz);
CHECK_ERROR_MT(playoutFreqHz);
- char bgnString[25];
- switch (*bgnMode) {
- case On: {
- *bgnMode = Fade;
- strncpy(bgnString, "Fade", 25);
- break;
- }
- case Fade: {
- *bgnMode = Off;
- strncpy(bgnString, "OFF", 25);
- break;
- }
- case Off: {
- *bgnMode = On;
- strncpy(bgnString, "ON", 25);
- break;
- }
- default:
- *bgnMode = On;
- strncpy(bgnString, "ON", 25);
- }
- CHECK_ERROR_MT(receiveACM->SetBackgroundNoiseMode(*bgnMode));
- bgnString[24] = '\0';
char playoutString[25];
switch (*playoutMode) {
@@ -1060,63 +1015,10 @@
fprintf(stdout, "Receive Frequency....... %d Hz\n", receiveFreqHz);
fprintf(stdout, "Playout Frequency....... %d Hz\n", playoutFreqHz);
fprintf(stdout, "Audio Playout Mode...... %s\n", playoutString);
- fprintf(stdout, "Background Noise Mode... %s\n", bgnString);
}
}
// set/get receiver VAD status & mode.
-void APITest::TestReceiverVAD(char side) {
- AudioCodingModule* myACM;
- int* myReceiveVADActivity;
-
- if (side == 'A') {
- myACM = _acmA;
- myReceiveVADActivity = _receiveVADActivityA;
- } else {
- myACM = _acmB;
- myReceiveVADActivity = _receiveVADActivityB;
- }
-
- ACMVADMode mode = myACM->ReceiveVADMode();
-
- CHECK_ERROR_MT(mode);
-
- if (!_randomTest) {
- fprintf(stdout, "\n\nCurrent Receive VAD at side %c\n", side);
- fprintf(stdout, "----------------------------------\n");
- fprintf(stdout, "mode.......... %d\n", (int) mode);
- fprintf(stdout, "VAD Active.... %d\n", myReceiveVADActivity[0]);
- fprintf(stdout, "VAD Passive... %d\n", myReceiveVADActivity[1]);
- fprintf(stdout, "VAD Unknown... %d\n", myReceiveVADActivity[2]);
- }
-
- if (!_randomTest) {
- fprintf(stdout, "\nChange Receive VAD at side %c\n\n", side);
- }
-
- switch (mode) {
- case VADNormal:
- mode = VADAggr;
- break;
- case VADLowBitrate:
- mode = VADVeryAggr;
- break;
- case VADAggr:
- mode = VADLowBitrate;
- break;
- case VADVeryAggr:
- mode = VADNormal;
- break;
- default:
- mode = VADNormal;
-
- CHECK_ERROR_MT(myACM->SetReceiveVADMode(mode));
- }
- for (int n = 0; n < 3; n++) {
- myReceiveVADActivity[n] = 0;
- }
-}
-
void APITest::TestSendVAD(char side) {
if (_randomTest) {
return;
@@ -1317,23 +1219,4 @@
Wait(500);
}
-void APITest::LookForDTMF(char side) {
- if (!_randomTest) {
- fprintf(stdout, "\n\nLooking for DTMF Signal in Side %c\n", side);
- fprintf(stdout, "----------------------------------------\n");
- }
-
- if (side == 'A') {
- _acmB->RegisterIncomingMessagesCallback(NULL);
- _acmA->RegisterIncomingMessagesCallback(_dtmfCallback);
- Wait(1000);
- _acmA->RegisterIncomingMessagesCallback(NULL);
- } else {
- _acmA->RegisterIncomingMessagesCallback(NULL);
- _acmB->RegisterIncomingMessagesCallback(_dtmfCallback);
- Wait(1000);
- _acmB->RegisterIncomingMessagesCallback(NULL);
- }
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/APITest.h b/webrtc/modules/audio_coding/main/test/APITest.h
index f29abf4..f9e9a91 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.h
+++ b/webrtc/modules/audio_coding/main/test/APITest.h
@@ -56,9 +56,6 @@
// Receiver Frequency, playout frequency.
void TestPlayout(char receiveSide);
- // set/get receiver VAD status & mode.
- void TestReceiverVAD(char side);
-
//
void TestSendVAD(char side);
@@ -68,8 +65,6 @@
void Wait(uint32_t waitLengthMs);
- void LookForDTMF(char side);
-
void RunTest(char thread);
bool PushAudioRunA();
@@ -145,11 +140,6 @@
AudioPlayoutMode _playoutModeA;
AudioPlayoutMode _playoutModeB;
- ACMBackgroundNoiseMode _bgnModeA;
- ACMBackgroundNoiseMode _bgnModeB;
-
- int _receiveVADActivityA[3];
- int _receiveVADActivityB[3];
bool _verbose;
int _dotPositionA;
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index bab207c..1ee6abc 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -20,7 +20,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index 620329b..29c9ade 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -16,7 +16,7 @@
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/system_wrappers/interface/trace.h"
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
index 57a912a..1a0f8f8 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -20,7 +20,7 @@
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
diff --git a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
index d8cdce5..85b1c8e 100644
--- a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "../source/acm_common_defs.h"
+#include "../acm2/acm_common_defs.h"
#include "gtest/gtest.h"
#include "audio_coding_module.h"
#include "PCMFile.h"
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index 50809fc..26f5b1f 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -23,7 +23,7 @@
#include <time.h>
#endif
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/main/test/utility.cc
index 62594ea..4b69640 100644
--- a/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/webrtc/modules/audio_coding/main/test/utility.cc
@@ -17,7 +17,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13