diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc | 3048 |
1 files changed, 0 insertions, 3048 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc deleted file mode 100644 index 556f530ecfd..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ /dev/null @@ -1,3048 +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/source/audio_coding_module_impl.h" - -#include <assert.h> -#include <stdlib.h> - -#include <algorithm> // For std::max. - -#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/acm2/call_statistics.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" -#include "webrtc/modules/audio_coding/main/acm2/nack.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/tick_util.h" -#include "webrtc/system_wrappers/interface/trace.h" -#include "webrtc/system_wrappers/interface/trace_event.h" - -namespace webrtc { - -namespace acm1 { - -enum { - kACMToneEnd = 999 -}; - -// Maximum number of bytes in one packet (PCM16B, 20 ms packets, stereo). -enum { - kMaxPacketSize = 2560 -}; - -// Maximum number of payloads that can be packed in one RED payload. For -// regular FEC, we only pack two payloads. In case of dual-streaming, in worst -// case we might pack 3 payloads in one RED payload. -enum { - kNumFecFragmentationVectors = 2, - kMaxNumFragmentationVectors = 3 -}; - -static const uint32_t kMaskTimestamp = 0x03ffffff; -static const int kDefaultTimestampDiff = 960; // 20 ms @ 48 kHz. - -// If packet N is arrived all packets prior to N - |kNackThresholdPackets| which -// are not received are considered as lost, and appear in NACK list. -static const int kNackThresholdPackets = 2; - -namespace { - -bool IsCodecRED(const CodecInst* codec) { - return (STR_CASE_CMP(codec->plname, "RED") == 0); -} - -bool IsCodecRED(int index) { - return (IsCodecRED(&ACMCodecDB::database_[index])); -} - -bool IsCodecCN(const CodecInst* codec) { - return (STR_CASE_CMP(codec->plname, "CN") == 0); -} - -bool IsCodecCN(int index) { - return (IsCodecCN(&ACMCodecDB::database_[index])); -} - -// Stereo-to-mono can be used as in-place. -int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { - if (length_out_buff < frame.samples_per_channel_) { - return -1; - } - for (int n = 0; n < frame.samples_per_channel_; ++n) - out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1; - return 0; -} - -// Mono-to-stereo can be used as in-place. -int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { - if (length_out_buff < frame.samples_per_channel_) { - return -1; - } - for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) { - out_buff[2 * n + 1] = frame.data_[n]; - out_buff[2 * n] = frame.data_[n]; - } - return 0; -} - -// Return 1 if timestamp t1 is less than timestamp t2, while compensating for -// wrap-around. -int TimestampLessThan(uint32_t t1, uint32_t t2) { - uint32_t kHalfFullRange = static_cast<uint32_t>(0xFFFFFFFF) / 2; - if (t1 == t2) { - return 0; - } else if (t1 < t2) { - if (t2 - t1 < kHalfFullRange) - return 1; - return 0; - } else { - if (t1 - t2 < kHalfFullRange) - return 0; - return 1; - } -} - -} // namespace - -AudioCodingModuleImpl::AudioCodingModuleImpl(const int32_t id, Clock* clock) - : packetization_callback_(NULL), - id_(id), - last_timestamp_(0xD87F3F9F), - last_in_timestamp_(0xD87F3F9F), - send_codec_inst_(), - cng_nb_pltype_(255), - cng_wb_pltype_(255), - cng_swb_pltype_(255), - cng_fb_pltype_(255), - red_pltype_(255), - vad_enabled_(false), - dtx_enabled_(false), - vad_mode_(VADNormal), - stereo_receive_registered_(false), - stereo_send_(false), - prev_received_channel_(0), - expected_channels_(1), - current_send_codec_idx_(-1), - current_receive_codec_idx_(-1), - send_codec_registered_(false), - acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - vad_callback_(NULL), - last_recv_audio_codec_pltype_(255), - is_first_red_(true), - fec_enabled_(false), - last_fec_timestamp_(0), - receive_red_pltype_(255), - previous_pltype_(255), - dummy_rtp_header_(NULL), - recv_pl_frame_size_smpls_(0), - receiver_initialized_(false), - dtmf_detector_(NULL), - dtmf_callback_(NULL), - last_detected_tone_(kACMToneEnd), - callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), - secondary_send_codec_inst_(), - initial_delay_ms_(0), - num_packets_accumulated_(0), - num_bytes_accumulated_(0), - accumulated_audio_ms_(0), - first_payload_received_(false), - last_incoming_send_timestamp_(0), - track_neteq_buffer_(false), - playout_ts_(0), - av_sync_(false), - last_timestamp_diff_(kDefaultTimestampDiff), - last_sequence_number_(0), - last_ssrc_(0), - last_packet_was_sync_(false), - clock_(clock), - nack_(), - nack_enabled_(false) { - - // Nullify send codec memory, set payload type and set codec name to - // invalid values. - const char no_name[] = "noCodecRegistered"; - strncpy(send_codec_inst_.plname, no_name, RTP_PAYLOAD_NAME_SIZE - 1); - send_codec_inst_.pltype = -1; - - strncpy(secondary_send_codec_inst_.plname, no_name, - RTP_PAYLOAD_NAME_SIZE - 1); - secondary_send_codec_inst_.pltype = -1; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - codecs_[i] = NULL; - registered_pltypes_[i] = -1; - stereo_receive_[i] = false; - slave_codecs_[i] = NULL; - mirror_codec_idx_[i] = -1; - } - - neteq_.set_id(id_); - - // Allocate memory for RED. - red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE]; - - // TODO(turajs): This might not be exactly how this class is supposed to work. - // The external usage might be that |fragmentationVectorSize| has to match - // the allocated space for the member-arrays, while here, we allocate - // according to the maximum number of fragmentations and change - // |fragmentationVectorSize| on-the-fly based on actual number of - // fragmentations. However, due to copying to local variable before calling - // SendData, the RTP module receives a "valid" fragmentation, where allocated - // space matches |fragmentationVectorSize|, therefore, this should not cause - // any problem. A better approach is not using RTPFragmentationHeader as - // member variable, instead, use an ACM-specific structure to hold RED-related - // data. See module_common_type.h for the definition of - // RTPFragmentationHeader. - fragmentation_.VerifyAndAllocateFragmentationHeader( - kMaxNumFragmentationVectors); - - // Register the default payload type for RED and for CNG at sampling rates of - // 8, 16, 32 and 48 kHz. - for (int i = (ACMCodecDB::kNumCodecs - 1); i >= 0; i--) { - if (IsCodecRED(i)) { - red_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (IsCodecCN(i)) { - if (ACMCodecDB::database_[i].plfreq == 8000) { - cng_nb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 16000) { - cng_wb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 32000) { - cng_swb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } else if (ACMCodecDB::database_[i].plfreq == 48000) { - cng_fb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); - } - } - } - - if (InitializeReceiverSafe() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot initialize receiver"); - } - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created"); -} - -AudioCodingModuleImpl::~AudioCodingModuleImpl() { - { - CriticalSectionScoped lock(acm_crit_sect_); - current_send_codec_idx_ = -1; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (codecs_[i] != NULL) { - // True stereo codecs share the same memory for master and - // slave, so slave codec need to be nullified here, since the - // memory will be deleted. - if (slave_codecs_[i] == codecs_[i]) { - slave_codecs_[i] = NULL; - } - - // Mirror index holds the address of the codec memory. - assert(mirror_codec_idx_[i] > -1); - if (codecs_[mirror_codec_idx_[i]] != NULL) { - delete codecs_[mirror_codec_idx_[i]]; - codecs_[mirror_codec_idx_[i]] = NULL; - } - - codecs_[i] = NULL; - } - - if (slave_codecs_[i] != NULL) { - // Delete memory for stereo usage of mono codecs. - assert(mirror_codec_idx_[i] > -1); - if (slave_codecs_[mirror_codec_idx_[i]] != NULL) { - delete slave_codecs_[mirror_codec_idx_[i]]; - slave_codecs_[mirror_codec_idx_[i]] = NULL; - } - slave_codecs_[i] = NULL; - } - } - - if (dtmf_detector_ != NULL) { - delete dtmf_detector_; - dtmf_detector_ = NULL; - } - if (dummy_rtp_header_ != NULL) { - delete dummy_rtp_header_; - dummy_rtp_header_ = NULL; - } - if (red_buffer_ != NULL) { - delete[] red_buffer_; - red_buffer_ = NULL; - } - } - - delete callback_crit_sect_; - callback_crit_sect_ = NULL; - - delete acm_crit_sect_; - acm_crit_sect_ = NULL; - WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, - "Destroyed"); -} - -int32_t AudioCodingModuleImpl::ChangeUniqueId(const int32_t id) { - { - CriticalSectionScoped lock(acm_crit_sect_); - id_ = id; - - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (codecs_[i] != NULL) { - codecs_[i]->SetUniqueID(id); - } - } - } - - neteq_.set_id(id_); - return 0; -} - -// Returns the number of milliseconds until the module want a -// worker thread to call Process. -int32_t AudioCodingModuleImpl::TimeUntilNextProcess() { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("TimeUntilNextProcess")) { - return -1; - } - return codecs_[current_send_codec_idx_]->SamplesLeftToEncode() / - (send_codec_inst_.plfreq / 1000); -} - -int32_t AudioCodingModuleImpl::Process() { - bool dual_stream; - { - CriticalSectionScoped lock(acm_crit_sect_); - dual_stream = (secondary_encoder_.get() != NULL); - } - if (dual_stream) { - return ProcessDualStream(); - } - return ProcessSingleStream(); -} - -int AudioCodingModuleImpl::EncodeFragmentation(int fragmentation_index, - int payload_type, - uint32_t current_timestamp, - ACMGenericCodec* encoder, - uint8_t* stream) { - int16_t len_bytes = MAX_PAYLOAD_SIZE_BYTE; - uint32_t rtp_timestamp; - WebRtcACMEncodingType encoding_type; - if (encoder->Encode(stream, &len_bytes, &rtp_timestamp, &encoding_type) < 0) { - return -1; - } - assert(encoding_type == kActiveNormalEncoded); - assert(len_bytes > 0); - - fragmentation_.fragmentationLength[fragmentation_index] = len_bytes; - fragmentation_.fragmentationPlType[fragmentation_index] = payload_type; - fragmentation_.fragmentationTimeDiff[fragmentation_index] = - static_cast<uint16_t>(current_timestamp - rtp_timestamp); - fragmentation_.fragmentationVectorSize++; - return len_bytes; -} - -// Primary payloads are sent immediately, whereas a single secondary payload is -// buffered to be combined with "the next payload." -// Normally "the next payload" would be a primary payload. In case two -// consecutive secondary payloads are generated with no primary payload in -// between, then two secondary payloads are packed in one RED. -int AudioCodingModuleImpl::ProcessDualStream() { - uint8_t stream[kMaxNumFragmentationVectors * MAX_PAYLOAD_SIZE_BYTE]; - uint32_t current_timestamp; - int16_t length_bytes = 0; - RTPFragmentationHeader my_fragmentation; - - uint8_t my_red_payload_type; - - { - CriticalSectionScoped lock(acm_crit_sect_); - // Check if there is an encoder before. - if (!HaveValidEncoder("ProcessDualStream") || - secondary_encoder_.get() == NULL) { - return -1; - } - ACMGenericCodec* primary_encoder = codecs_[current_send_codec_idx_]; - // If primary encoder has a full frame of audio to generate payload. - bool primary_ready_to_encode = primary_encoder->HasFrameToEncode(); - // If the secondary encoder has a frame of audio to generate a payload. - bool secondary_ready_to_encode = secondary_encoder_->HasFrameToEncode(); - - if (!primary_ready_to_encode && !secondary_ready_to_encode) { - // Nothing to send. - return 0; - } - int len_bytes_previous_secondary = static_cast<int>( - fragmentation_.fragmentationLength[2]); - assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE); - bool has_previous_payload = len_bytes_previous_secondary > 0; - - uint32_t primary_timestamp = primary_encoder->EarliestTimestamp(); - uint32_t secondary_timestamp = secondary_encoder_->EarliestTimestamp(); - - if (!has_previous_payload && !primary_ready_to_encode && - secondary_ready_to_encode) { - // Secondary payload will be the ONLY bit-stream. Encode by secondary - // encoder, store the payload, and return. No packet is sent. - int16_t len_bytes = MAX_PAYLOAD_SIZE_BYTE; - WebRtcACMEncodingType encoding_type; - if (secondary_encoder_->Encode(red_buffer_, &len_bytes, - &last_fec_timestamp_, - &encoding_type) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDual(): Encoding of secondary encoder Failed"); - return -1; - } - assert(len_bytes > 0); - assert(encoding_type == kActiveNormalEncoded); - assert(len_bytes <= MAX_PAYLOAD_SIZE_BYTE); - fragmentation_.fragmentationLength[2] = len_bytes; - return 0; - } - - // Initialize with invalid but different values, so later can have sanity - // check if they are different. - int index_primary = -1; - int index_secondary = -2; - int index_previous_secondary = -3; - - if (primary_ready_to_encode) { - index_primary = secondary_ready_to_encode ? - TimestampLessThan(primary_timestamp, secondary_timestamp) : 0; - index_primary += has_previous_payload ? - TimestampLessThan(primary_timestamp, last_fec_timestamp_) : 0; - } - - if (secondary_ready_to_encode) { - // Timestamp of secondary payload can only be less than primary payload, - // but is always larger than the timestamp of previous secondary payload. - index_secondary = primary_ready_to_encode ? - (1 - TimestampLessThan(primary_timestamp, secondary_timestamp)) : 0; - } - - if (has_previous_payload) { - index_previous_secondary = primary_ready_to_encode ? - (1 - TimestampLessThan(primary_timestamp, last_fec_timestamp_)) : 0; - // If secondary is ready it always have a timestamp larger than previous - // secondary. So the index is either 0 or 1. - index_previous_secondary += secondary_ready_to_encode ? 1 : 0; - } - - // Indices must not be equal. - assert(index_primary != index_secondary); - assert(index_primary != index_previous_secondary); - assert(index_secondary != index_previous_secondary); - - // One of the payloads has to be at position zero. - assert(index_primary == 0 || index_secondary == 0 || - index_previous_secondary == 0); - - // Timestamp of the RED payload. - if (index_primary == 0) { - current_timestamp = primary_timestamp; - } else if (index_secondary == 0) { - current_timestamp = secondary_timestamp; - } else { - current_timestamp = last_fec_timestamp_; - } - - fragmentation_.fragmentationVectorSize = 0; - if (has_previous_payload) { - assert(index_previous_secondary >= 0 && - index_previous_secondary < kMaxNumFragmentationVectors); - assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE); - memcpy(&stream[index_previous_secondary * MAX_PAYLOAD_SIZE_BYTE], - red_buffer_, sizeof(stream[0]) * len_bytes_previous_secondary); - fragmentation_.fragmentationLength[index_previous_secondary] = - len_bytes_previous_secondary; - fragmentation_.fragmentationPlType[index_previous_secondary] = - secondary_send_codec_inst_.pltype; - fragmentation_.fragmentationTimeDiff[index_previous_secondary] = - static_cast<uint16_t>(current_timestamp - last_fec_timestamp_); - fragmentation_.fragmentationVectorSize++; - } - - if (primary_ready_to_encode) { - assert(index_primary >= 0 && index_primary < kMaxNumFragmentationVectors); - int i = index_primary * MAX_PAYLOAD_SIZE_BYTE; - if (EncodeFragmentation(index_primary, send_codec_inst_.pltype, - current_timestamp, primary_encoder, - &stream[i]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDualStream(): Encoding of primary encoder Failed"); - return -1; - } - } - - if (secondary_ready_to_encode) { - assert(index_secondary >= 0 && - index_secondary < kMaxNumFragmentationVectors - 1); - int i = index_secondary * MAX_PAYLOAD_SIZE_BYTE; - if (EncodeFragmentation(index_secondary, - secondary_send_codec_inst_.pltype, - current_timestamp, secondary_encoder_.get(), - &stream[i]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessDualStream(): Encoding of secondary encoder " - "Failed"); - return -1; - } - } - // Copy to local variable, as it will be used outside the ACM lock. - my_fragmentation.CopyFrom(fragmentation_); - my_red_payload_type = red_pltype_; - length_bytes = 0; - for (int n = 0; n < fragmentation_.fragmentationVectorSize; n++) { - length_bytes += fragmentation_.fragmentationLength[n]; - } - } - - { - CriticalSectionScoped lock(callback_crit_sect_); - if (packetization_callback_ != NULL) { - // Callback with payload data, including redundant data (FEC/RED). - if (packetization_callback_->SendData(kAudioFrameSpeech, - my_red_payload_type, - current_timestamp, stream, - length_bytes, - &my_fragmentation) < 0) { - return -1; - } - } - } - - { - CriticalSectionScoped lock(acm_crit_sect_); - // Now that data is sent, clean up fragmentation. - ResetFragmentation(0); - } - return 0; -} - -// Process any pending tasks such as timeouts. -int AudioCodingModuleImpl::ProcessSingleStream() { - // Make room for 1 RED payload. - uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE]; - int16_t length_bytes = 2 * MAX_PAYLOAD_SIZE_BYTE; - int16_t red_length_bytes = length_bytes; - uint32_t rtp_timestamp; - int16_t status; - WebRtcACMEncodingType encoding_type; - FrameType frame_type = kAudioFrameSpeech; - uint8_t current_payload_type = 0; - bool has_data_to_send = false; - bool fec_active = false; - RTPFragmentationHeader my_fragmentation; - - // Keep the scope of the ACM critical section limited. - { - CriticalSectionScoped lock(acm_crit_sect_); - // Check if there is an encoder before. - if (!HaveValidEncoder("ProcessSingleStream")) { - return -1; - } - status = codecs_[current_send_codec_idx_]->Encode(stream, &length_bytes, - &rtp_timestamp, - &encoding_type); - if (status < 0) { - // Encode failed. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ProcessSingleStream(): Encoding Failed"); - length_bytes = 0; - return -1; - } else if (status == 0) { - // Not enough data. - return 0; - } else { - switch (encoding_type) { - case kNoEncoding: { - current_payload_type = previous_pltype_; - frame_type = kFrameEmpty; - length_bytes = 0; - break; - } - case kActiveNormalEncoded: - case kPassiveNormalEncoded: { - current_payload_type = static_cast<uint8_t>(send_codec_inst_.pltype); - frame_type = kAudioFrameSpeech; - break; - } - case kPassiveDTXNB: { - current_payload_type = cng_nb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXWB: { - current_payload_type = cng_wb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXSWB: { - current_payload_type = cng_swb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - case kPassiveDTXFB: { - current_payload_type = cng_fb_pltype_; - frame_type = kAudioFrameCN; - is_first_red_ = true; - break; - } - } - has_data_to_send = true; - previous_pltype_ = current_payload_type; - - // Redundancy encode is done here. The two bitstreams packetized into - // one RTP packet and the fragmentation points are set. - // Only apply RED on speech data. - if ((fec_enabled_) && - ((encoding_type == kActiveNormalEncoded) || - (encoding_type == kPassiveNormalEncoded))) { - // FEC is enabled within this scope. - // - // Note that, a special solution exists for iSAC since it is the only - // codec for which GetRedPayload has a non-empty implementation. - // - // Summary of the FEC scheme below (use iSAC as example): - // - // 1st (is_first_red_ is true) encoded iSAC frame (primary #1) => - // - call GetRedPayload() and store redundancy for packet #1 in - // second fragment of RED buffer (old data) - // - drop the primary iSAC frame - // - don't call SendData - // 2nd (is_first_red_ is false) encoded iSAC frame (primary #2) => - // - store primary #2 in 1st fragment of RED buffer and send the - // combined packet - // - the transmitted packet contains primary #2 (new) and - // reduncancy for packet #1 (old) - // - call GetRed_Payload() and store redundancy for packet #2 in - // second fragment of RED buffer - // - // ... - // - // Nth encoded iSAC frame (primary #N) => - // - store primary #N in 1st fragment of RED buffer and send the - // combined packet - // - the transmitted packet contains primary #N (new) and - // reduncancy for packet #(N-1) (old) - // - call GetRedPayload() and store redundancy for packet #N in - // second fragment of RED buffer - // - // For all other codecs, GetRedPayload does nothing and returns -1 => - // redundant data is only a copy. - // - // First combined packet contains : #2 (new) and #1 (old) - // Second combined packet contains: #3 (new) and #2 (old) - // Third combined packet contains : #4 (new) and #3 (old) - // - // Hence, even if every second packet is dropped, perfect - // reconstruction is possible. - fec_active = true; - - has_data_to_send = false; - // Skip the following part for the first packet in a RED session. - if (!is_first_red_) { - // Rearrange stream such that FEC packets are included. - // Replace stream now that we have stored current stream. - memcpy(stream + fragmentation_.fragmentationOffset[1], red_buffer_, - fragmentation_.fragmentationLength[1]); - // Update the fragmentation time difference vector, in number of - // timestamps. - uint16_t time_since_last = static_cast<uint16_t>(rtp_timestamp - - last_fec_timestamp_); - - // Update fragmentation vectors. - fragmentation_.fragmentationPlType[1] = - fragmentation_.fragmentationPlType[0]; - fragmentation_.fragmentationTimeDiff[1] = time_since_last; - has_data_to_send = true; - } - - // Insert new packet length. - fragmentation_.fragmentationLength[0] = length_bytes; - - // Insert new packet payload type. - fragmentation_.fragmentationPlType[0] = current_payload_type; - last_fec_timestamp_ = rtp_timestamp; - - // Can be modified by the GetRedPayload() call if iSAC is utilized. - red_length_bytes = length_bytes; - - // A fragmentation header is provided => packetization according to - // RFC 2198 (RTP Payload for Redundant Audio Data) will be used. - // First fragment is the current data (new). - // Second fragment is the previous data (old). - length_bytes = static_cast<int16_t>( - fragmentation_.fragmentationLength[0] + - fragmentation_.fragmentationLength[1]); - - // Get, and store, redundant data from the encoder based on the recently - // encoded frame. - // NOTE - only iSAC contains an implementation; all other codecs does - // nothing and returns -1. - if (codecs_[current_send_codec_idx_]->GetRedPayload( - red_buffer_, - &red_length_bytes) == -1) { - // The codec was not iSAC => use current encoder output as redundant - // data instead (trivial FEC scheme). - memcpy(red_buffer_, stream, red_length_bytes); - } - - is_first_red_ = false; - // Update payload type with RED payload type. - current_payload_type = red_pltype_; - // We have packed 2 payloads. - fragmentation_.fragmentationVectorSize = kNumFecFragmentationVectors; - - // Copy to local variable, as it will be used outside ACM lock. - my_fragmentation.CopyFrom(fragmentation_); - // Store RED length. - fragmentation_.fragmentationLength[1] = red_length_bytes; - } - } - } - - if (has_data_to_send) { - CriticalSectionScoped lock(callback_crit_sect_); - - if (packetization_callback_ != NULL) { - if (fec_active) { - // Callback with payload data, including redundant data (FEC/RED). - packetization_callback_->SendData(frame_type, current_payload_type, - rtp_timestamp, stream, - length_bytes, - &my_fragmentation); - } else { - // Callback with payload data. - packetization_callback_->SendData(frame_type, current_payload_type, - rtp_timestamp, stream, - length_bytes, NULL); - } - } - - if (vad_callback_ != NULL) { - // Callback with VAD decision. - vad_callback_->InFrameType(static_cast<int16_t>(encoding_type)); - } - } - return length_bytes; -} - -///////////////////////////////////////// -// Sender -// - -// Initialize send codec. -int32_t AudioCodingModuleImpl::InitializeSender() { - CriticalSectionScoped lock(acm_crit_sect_); - - // Start with invalid values. - send_codec_registered_ = false; - current_send_codec_idx_ = -1; - send_codec_inst_.plname[0] = '\0'; - - // Delete all encoders to start fresh. - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (codecs_[id] != NULL) { - codecs_[id]->DestructEncoder(); - } - } - - // Initialize FEC/RED. - is_first_red_ = true; - if (fec_enabled_ || secondary_encoder_.get() != NULL) { - if (red_buffer_ != NULL) { - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - } - if (fec_enabled_) { - ResetFragmentation(kNumFecFragmentationVectors); - } else { - ResetFragmentation(0); - } - } - - return 0; -} - -int32_t AudioCodingModuleImpl::ResetEncoder() { - CriticalSectionScoped lock(acm_crit_sect_); - if (!HaveValidEncoder("ResetEncoder")) { - return -1; - } - return codecs_[current_send_codec_idx_]->ResetEncoder(); -} - -void AudioCodingModuleImpl::UnregisterSendCodec() { - CriticalSectionScoped lock(acm_crit_sect_); - send_codec_registered_ = false; - current_send_codec_idx_ = -1; - // If send Codec is unregistered then remove the secondary codec as well. - if (secondary_encoder_.get() != NULL) - secondary_encoder_.reset(); - return; -} - -ACMGenericCodec* AudioCodingModuleImpl::CreateCodec(const CodecInst& codec) { - ACMGenericCodec* my_codec = NULL; - - my_codec = ACMCodecDB::CreateCodecInstance(&codec); - if (my_codec == NULL) { - // Error, could not create the codec. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ACMCodecDB::CreateCodecInstance() failed in CreateCodec()"); - return my_codec; - } - my_codec->SetUniqueID(id_); - my_codec->SetNetEqDecodeLock(neteq_.DecodeLock()); - - return my_codec; -} - -// Check if the given codec is a valid to be registered as send codec. -static int IsValidSendCodec(const CodecInst& send_codec, - bool is_primary_encoder, - int acm_id, - int* mirror_id) { - if ((send_codec.channels != 1) && (send_codec.channels != 2)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Wrong number of channels (%d, only mono and stereo are " - "supported) for %s encoder", send_codec.channels, - is_primary_encoder ? "primary" : "secondary"); - return -1; - } - - int codec_id = ACMCodecDB::CodecNumber(&send_codec, mirror_id); - if (codec_id < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Invalid settings for the send codec."); - return -1; - } - - // TODO(tlegrand): Remove this check. Already taken care of in - // ACMCodecDB::CodecNumber(). - // Check if the payload-type is valid - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "Invalid payload-type %d for %s.", send_codec.pltype, - send_codec.plname); - return -1; - } - - // Telephone-event cannot be a send codec. - if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "telephone-event cannot be a send codec"); - *mirror_id = -1; - return -1; - } - - if (ACMCodecDB::codec_settings_[codec_id].channel_support - < send_codec.channels) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "%d number of channels not supportedn for %s.", - send_codec.channels, send_codec.plname); - *mirror_id = -1; - return -1; - } - - if (!is_primary_encoder) { - // If registering the secondary encoder, then RED and CN are not valid - // choices as encoder. - if (IsCodecRED(&send_codec)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "RED cannot be secondary codec"); - *mirror_id = -1; - return -1; - } - - if (IsCodecCN(&send_codec)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, acm_id, - "DTX cannot be secondary codec"); - *mirror_id = -1; - return -1; - } - } - return codec_id; -} - -int AudioCodingModuleImpl::RegisterSecondarySendCodec( - const CodecInst& send_codec) { - CriticalSectionScoped lock(acm_crit_sect_); - if (!send_codec_registered_) { - return -1; - } - // Primary and Secondary codecs should have the same sampling rates. - if (send_codec.plfreq != send_codec_inst_.plfreq) { - return -1; - } - int mirror_id; - int codec_id = IsValidSendCodec(send_codec, false, id_, &mirror_id); - if (codec_id < 0) { - return -1; - } - ACMGenericCodec* encoder = CreateCodec(send_codec); - WebRtcACMCodecParams codec_params; - // Initialize the codec before registering. For secondary codec VAD & DTX are - // disabled. - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = false; - codec_params.enable_dtx = false; - codec_params.vad_mode = VADNormal; - // Force initialization. - if (encoder->InitEncoder(&codec_params, true) < 0) { - // Could not initialize, therefore cannot be registered. - delete encoder; - return -1; - } - secondary_encoder_.reset(encoder); - memcpy(&secondary_send_codec_inst_, &send_codec, sizeof(send_codec)); - - // Disable VAD & DTX. - SetVADSafe(false, false, VADNormal); - - // Cleaning. - if (red_buffer_) { - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - } - ResetFragmentation(0); - return 0; -} - -void AudioCodingModuleImpl::UnregisterSecondarySendCodec() { - CriticalSectionScoped lock(acm_crit_sect_); - if (secondary_encoder_.get() == NULL) { - return; - } - secondary_encoder_.reset(); - ResetFragmentation(0); -} - -int AudioCodingModuleImpl::SecondarySendCodec( - CodecInst* secondary_codec) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (secondary_encoder_.get() == NULL) { - return -1; - } - memcpy(secondary_codec, &secondary_send_codec_inst_, - sizeof(secondary_send_codec_inst_)); - return 0; -} - -// Can be called multiple times for Codec, CNG, RED. -int32_t AudioCodingModuleImpl::RegisterSendCodec( - const CodecInst& send_codec) { - int mirror_id; - int codec_id = IsValidSendCodec(send_codec, true, id_, &mirror_id); - - CriticalSectionScoped lock(acm_crit_sect_); - - // Check for reported errors from function IsValidSendCodec(). - if (codec_id < 0) { - if (!send_codec_registered_) { - // This values has to be NULL if there is no codec registered. - current_send_codec_idx_ = -1; - } - return -1; - } - - // RED can be registered with other payload type. If not registered a default - // payload type is used. - if (IsCodecRED(&send_codec)) { - // TODO(tlegrand): Remove this check. Already taken care of in - // ACMCodecDB::CodecNumber(). - // Check if the payload-type is valid - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid payload-type %d for %s.", send_codec.pltype, - send_codec.plname); - return -1; - } - // Set RED payload type. - red_pltype_ = static_cast<uint8_t>(send_codec.pltype); - return 0; - } - - // CNG can be registered with other payload type. If not registered the - // default payload types from codec database will be used. - if (IsCodecCN(&send_codec)) { - // CNG is registered. - switch (send_codec.plfreq) { - case 8000: { - cng_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 16000: { - cng_wb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 32000: { - cng_swb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - case 48000: { - cng_fb_pltype_ = static_cast<uint8_t>(send_codec.pltype); - break; - } - default: { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RegisterSendCodec() failed, invalid frequency for CNG " - "registration"); - return -1; - } - } - return 0; - } - - // Set Stereo, and make sure VAD and DTX is turned off. - if (send_codec.channels == 2) { - stereo_send_ = true; - if (vad_enabled_ || dtx_enabled_) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "VAD/DTX is turned off, not supported when sending stereo."); - } - vad_enabled_ = false; - dtx_enabled_ = false; - } else { - stereo_send_ = false; - } - - // Check if the codec is already registered as send codec. - bool is_send_codec; - if (send_codec_registered_) { - int send_codec_mirror_id; - int send_codec_id = ACMCodecDB::CodecNumber(&send_codec_inst_, - &send_codec_mirror_id); - assert(send_codec_id >= 0); - is_send_codec = (send_codec_id == codec_id) || - (mirror_id == send_codec_mirror_id); - } else { - is_send_codec = false; - } - - // If there is secondary codec registered and the new send codec has a - // sampling rate different than that of secondary codec, then unregister the - // secondary codec. - if (secondary_encoder_.get() != NULL && - secondary_send_codec_inst_.plfreq != send_codec.plfreq) { - secondary_encoder_.reset(); - ResetFragmentation(0); - } - - // If new codec, or new settings, register. - if (!is_send_codec) { - if (codecs_[mirror_id] == NULL) { - codecs_[mirror_id] = CreateCodec(send_codec); - if (codecs_[mirror_id] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Create the codec"); - return -1; - } - mirror_codec_idx_[mirror_id] = mirror_id; - } - - if (mirror_id != codec_id) { - codecs_[codec_id] = codecs_[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - ACMGenericCodec* codec_ptr = codecs_[codec_id]; - WebRtcACMCodecParams codec_params; - - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = vad_enabled_; - codec_params.enable_dtx = dtx_enabled_; - codec_params.vad_mode = vad_mode_; - // Force initialization. - if (codec_ptr->InitEncoder(&codec_params, true) < 0) { - // Could not initialize the encoder. - - // Check if already have a registered codec. - // Depending on that different messages are logged. - if (!send_codec_registered_) { - current_send_codec_idx_ = -1; - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Initialize the encoder No Encoder is registered"); - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Initialize the encoder, continue encoding with " - "the previously registered codec"); - } - return -1; - } - - // Update states. - dtx_enabled_ = codec_params.enable_dtx; - vad_enabled_ = codec_params.enable_vad; - vad_mode_ = codec_params.vad_mode; - - // Everything is fine so we can replace the previous codec with this one. - if (send_codec_registered_) { - // If we change codec we start fresh with FEC. - // This is not strictly required by the standard. - is_first_red_ = true; - - codec_ptr->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_); - } - - current_send_codec_idx_ = codec_id; - send_codec_registered_ = true; - memcpy(&send_codec_inst_, &send_codec, sizeof(CodecInst)); - previous_pltype_ = send_codec_inst_.pltype; - return 0; - } else { - // If codec is the same as already registered check if any parameters - // has changed compared to the current values. - // If any parameter is valid then apply it and record. - bool force_init = false; - - if (mirror_id != codec_id) { - codecs_[codec_id] = codecs_[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - // Check the payload type. - if (send_codec.pltype != send_codec_inst_.pltype) { - // At this point check if the given payload type is valid. - // Record it later when the sampling frequency is changed - // successfully. - if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Out of range payload type"); - return -1; - } - } - - // If there is a codec that ONE instance of codec supports multiple - // sampling frequencies, then we need to take care of it here. - // one such a codec is iSAC. Both WB and SWB are encoded and decoded - // with one iSAC instance. Therefore, we need to update the encoder - // frequency if required. - if (send_codec_inst_.plfreq != send_codec.plfreq) { - force_init = true; - - // If sampling frequency is changed we have to start fresh with RED. - is_first_red_ = true; - } - - // If packet size or number of channels has changed, we need to - // re-initialize the encoder. - if (send_codec_inst_.pacsize != send_codec.pacsize) { - force_init = true; - } - if (send_codec_inst_.channels != send_codec.channels) { - force_init = true; - } - - if (force_init) { - WebRtcACMCodecParams codec_params; - - memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst)); - codec_params.enable_vad = vad_enabled_; - codec_params.enable_dtx = dtx_enabled_; - codec_params.vad_mode = vad_mode_; - - // Force initialization. - if (codecs_[current_send_codec_idx_]->InitEncoder(&codec_params, - true) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Could not change the codec packet-size."); - return -1; - } - - send_codec_inst_.plfreq = send_codec.plfreq; - send_codec_inst_.pacsize = send_codec.pacsize; - send_codec_inst_.channels = send_codec.channels; - } - - // If the change of sampling frequency has been successful then - // we store the payload-type. - send_codec_inst_.pltype = send_codec.pltype; - - // Check if a change in Rate is required. - if (send_codec.rate != send_codec_inst_.rate) { - if (codecs_[codec_id]->SetBitRate(send_codec.rate) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Could not change the codec rate."); - return -1; - } - send_codec_inst_.rate = send_codec.rate; - } - previous_pltype_ = send_codec_inst_.pltype; - - return 0; - } -} - -// Get current send codec. -int32_t AudioCodingModuleImpl::SendCodec( - CodecInst* current_codec) const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendCodec()"); - CriticalSectionScoped lock(acm_crit_sect_); - - assert(current_codec); - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendCodec Failed, no codec is registered"); - - return -1; - } - WebRtcACMCodecParams encoder_param; - codecs_[current_send_codec_idx_]->EncoderParams(&encoder_param); - encoder_param.codec_inst.pltype = send_codec_inst_.pltype; - memcpy(current_codec, &(encoder_param.codec_inst), sizeof(CodecInst)); - - return 0; -} - -// Get current send frequency. -int32_t AudioCodingModuleImpl::SendFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendFrequency()"); - CriticalSectionScoped lock(acm_crit_sect_); - - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendFrequency Failed, no codec is registered"); - - return -1; - } - - return send_codec_inst_.plfreq; -} - -// Get encode bitrate. -// Adaptive rate codecs return their current encode target rate, while other -// codecs return there longterm avarage or their fixed rate. -int32_t AudioCodingModuleImpl::SendBitrate() const { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!send_codec_registered_) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "SendBitrate Failed, no codec is registered"); - - return -1; - } - - WebRtcACMCodecParams encoder_param; - codecs_[current_send_codec_idx_]->EncoderParams(&encoder_param); - - return encoder_param.codec_inst.rate; -} - -// Set available bandwidth, inform the encoder about the estimated bandwidth -// received from the remote party. -int32_t AudioCodingModuleImpl::SetReceivedEstimatedBandwidth( - const int32_t bw) { - return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw); -} - -// Register a transport callback which will be called to deliver -// the encoded buffers. -int32_t AudioCodingModuleImpl::RegisterTransportCallback( - AudioPacketizationCallback* transport) { - CriticalSectionScoped lock(callback_crit_sect_); - packetization_callback_ = transport; - return 0; -} - -// Add 10MS of raw (PCM) audio data to the encoder. -int32_t AudioCodingModuleImpl::Add10MsData( - const AudioFrame& audio_frame) { - if (audio_frame.samples_per_channel_ <= 0) { - assert(false); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, payload length is negative or " - "zero"); - return -1; - } - - // Allow for 8, 16, 32 and 48kHz input audio. - if ((audio_frame.sample_rate_hz_ != 8000) - && (audio_frame.sample_rate_hz_ != 16000) - && (audio_frame.sample_rate_hz_ != 32000) - && (audio_frame.sample_rate_hz_ != 48000)) { - assert(false); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, input frequency not valid"); - return -1; - } - - // If the length and frequency matches. We currently just support raw PCM. - if ((audio_frame.sample_rate_hz_ / 100) - != audio_frame.samples_per_channel_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, input frequency and length doesn't" - " match"); - return -1; - } - - if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot Add 10 ms audio, invalid number of channels."); - return -1; - } - - CriticalSectionScoped lock(acm_crit_sect_); - // Do we have a codec registered? - if (!HaveValidEncoder("Add10MsData")) { - return -1; - } - - const AudioFrame* ptr_frame; - // Perform a resampling, also down-mix if it is required and can be - // performed before resampling (a down mix prior to resampling will take - // place if both primary and secondary encoders are mono and input is in - // stereo). - if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) { - return -1; - } - TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Audio", ptr_frame->timestamp_, - "now", clock_->TimeInMilliseconds()); - - // Check whether we need an up-mix or down-mix? - bool remix = ptr_frame->num_channels_ != send_codec_inst_.channels; - if (secondary_encoder_.get() != NULL) { - remix = remix || - (ptr_frame->num_channels_ != secondary_send_codec_inst_.channels); - } - - // If a re-mix is required (up or down), this buffer will store re-mixed - // version of the input. - int16_t buffer[WEBRTC_10MS_PCM_AUDIO]; - if (remix) { - if (ptr_frame->num_channels_ == 1) { - if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, buffer) < 0) - return -1; - } else { - if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, buffer) < 0) - return -1; - } - } - - // When adding data to encoders this pointer is pointing to an audio buffer - // with correct number of channels. - const int16_t* ptr_audio = ptr_frame->data_; - - // For pushing data to primary, point the |ptr_audio| to correct buffer. - if (send_codec_inst_.channels != ptr_frame->num_channels_) - ptr_audio = buffer; - - if (codecs_[current_send_codec_idx_]->Add10MsData( - ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, - send_codec_inst_.channels) < 0) - return -1; - - if (secondary_encoder_.get() != NULL) { - // For pushing data to secondary, point the |ptr_audio| to correct buffer. - ptr_audio = ptr_frame->data_; - if (secondary_send_codec_inst_.channels != ptr_frame->num_channels_) - ptr_audio = buffer; - - if (secondary_encoder_->Add10MsData( - ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, - secondary_send_codec_inst_.channels) < 0) - return -1; - } - - return 0; -} - -// Perform a resampling and down-mix if required. We down-mix only if -// encoder is mono and input is stereo. In case of dual-streaming, both -// encoders has to be mono for down-mix to take place. -// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing -// is required, |*ptr_out| points to |in_frame|. -int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, - const AudioFrame** ptr_out) { - // Primary and secondary (if exists) should have the same sampling rate. - assert((secondary_encoder_.get() != NULL) ? - secondary_send_codec_inst_.plfreq == send_codec_inst_.plfreq : true); - - bool resample = static_cast<int32_t>(in_frame.sample_rate_hz_) != - send_codec_inst_.plfreq; - - // This variable is true if primary codec and secondary codec (if exists) - // are both mono and input is stereo. - bool down_mix; - if (secondary_encoder_.get() != NULL) { - down_mix = (in_frame.num_channels_ == 2) && - (send_codec_inst_.channels == 1) && - (secondary_send_codec_inst_.channels == 1); - } else { - down_mix = (in_frame.num_channels_ == 2) && - (send_codec_inst_.channels == 1); - } - - if (!down_mix && !resample) { - // No pre-processing is required. - last_in_timestamp_ = in_frame.timestamp_; - last_timestamp_ = in_frame.timestamp_; - *ptr_out = &in_frame; - return 0; - } - - *ptr_out = &preprocess_frame_; - preprocess_frame_.num_channels_ = in_frame.num_channels_; - int16_t audio[WEBRTC_10MS_PCM_AUDIO]; - const int16_t* src_ptr_audio = in_frame.data_; - int16_t* dest_ptr_audio = preprocess_frame_.data_; - if (down_mix) { - // If a resampling is required the output of a down-mix is written into a - // local buffer, otherwise, it will be written to the output frame. - if (resample) - dest_ptr_audio = audio; - if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0) - return -1; - preprocess_frame_.num_channels_ = 1; - // Set the input of the resampler is the down-mixed signal. - src_ptr_audio = audio; - } - - preprocess_frame_.timestamp_ = in_frame.timestamp_; - preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; - preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; - // If it is required, we have to do a resampling. - if (resample) { - // The result of the resampler is written to output frame. - dest_ptr_audio = preprocess_frame_.data_; - - uint32_t timestamp_diff; - - // Calculate the timestamp of this frame. - if (last_in_timestamp_ > in_frame.timestamp_) { - // A wrap around has happened. - timestamp_diff = (static_cast<uint32_t>(0xFFFFFFFF) - last_in_timestamp_) - + in_frame.timestamp_; - } else { - timestamp_diff = in_frame.timestamp_ - last_in_timestamp_; - } - preprocess_frame_.timestamp_ = last_timestamp_ + - static_cast<uint32_t>(timestamp_diff * - (static_cast<double>(send_codec_inst_.plfreq) / - static_cast<double>(in_frame.sample_rate_hz_))); - - preprocess_frame_.samples_per_channel_ = input_resampler_.Resample10Msec( - src_ptr_audio, in_frame.sample_rate_hz_, dest_ptr_audio, - send_codec_inst_.plfreq, preprocess_frame_.num_channels_); - - if (preprocess_frame_.samples_per_channel_ < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add 10 ms audio, resmapling failed"); - return -1; - } - preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq; - } - last_in_timestamp_ = in_frame.timestamp_; - last_timestamp_ = preprocess_frame_.timestamp_; - - return 0; -} - -///////////////////////////////////////// -// (FEC) Forward Error Correction -// - -bool AudioCodingModuleImpl::FECStatus() const { - CriticalSectionScoped lock(acm_crit_sect_); - return fec_enabled_; -} - -// Configure FEC status i.e on/off. -int32_t -AudioCodingModuleImpl::SetFECStatus( -#ifdef WEBRTC_CODEC_RED - const bool enable_fec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (fec_enabled_ != enable_fec) { - // Reset the RED buffer. - memset(red_buffer_, 0, MAX_PAYLOAD_SIZE_BYTE); - - // Reset fragmentation buffers. - ResetFragmentation(kNumFecFragmentationVectors); - // Set fec_enabled_. - fec_enabled_ = enable_fec; - } - is_first_red_ = true; // Make sure we restart FEC. - return 0; -#else - const bool /* enable_fec */) { - fec_enabled_ = false; - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - " WEBRTC_CODEC_RED is undefined => fec_enabled_ = %d", - fec_enabled_); - return -1; -#endif -} - -///////////////////////////////////////// -// (VAD) Voice Activity Detection -// -int32_t AudioCodingModuleImpl::SetVAD(bool enable_dtx, bool enable_vad, - ACMVADMode mode) { - CriticalSectionScoped lock(acm_crit_sect_); - return SetVADSafe(enable_dtx, enable_vad, mode); -} - -int AudioCodingModuleImpl::SetVADSafe(bool enable_dtx, bool enable_vad, - ACMVADMode mode) { - // Sanity check of the mode. - if ((mode != VADNormal) && (mode != VADLowBitrate) - && (mode != VADAggr) && (mode != VADVeryAggr)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid VAD Mode %d, no change is made to VAD/DTX status", - static_cast<int>(mode)); - return -1; - } - - // Check that the send codec is mono. We don't support VAD/DTX for stereo - // sending. - if ((enable_dtx || enable_vad) && stereo_send_) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "VAD/DTX not supported for stereo sending."); - dtx_enabled_ = false; - vad_enabled_ = false; - vad_mode_ = mode; - return -1; - } - - // We don't support VAD/DTX when dual-streaming is enabled, i.e. - // secondary-encoder is registered. - if ((enable_dtx || enable_vad) && secondary_encoder_.get() != NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "VAD/DTX not supported when dual-streaming is enabled."); - dtx_enabled_ = false; - vad_enabled_ = false; - vad_mode_ = mode; - return -1; - } - - // Store VAD/DTX settings. Values can be changed in the call to "SetVAD" - // below. - dtx_enabled_ = enable_dtx; - vad_enabled_ = enable_vad; - vad_mode_ = mode; - - // If a send codec is registered, set VAD/DTX for the codec. - if (HaveValidEncoder("SetVAD")) { - if (codecs_[current_send_codec_idx_]->SetVAD(&dtx_enabled_, &vad_enabled_, - &vad_mode_) < 0) { - // SetVAD failed. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "SetVAD failed"); - dtx_enabled_ = false; - vad_enabled_ = false; - return -1; - } - } - - return 0; -} - -// Get VAD/DTX settings. -// TODO(tlegrand): Change this method to void. -int32_t AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled, - ACMVADMode* mode) const { - CriticalSectionScoped lock(acm_crit_sect_); - - *dtx_enabled = dtx_enabled_; - *vad_enabled = vad_enabled_; - *mode = vad_mode_; - - return 0; -} - -///////////////////////////////////////// -// Receiver -// - -int32_t AudioCodingModuleImpl::InitializeReceiver() { - CriticalSectionScoped lock(acm_crit_sect_); - return InitializeReceiverSafe(); -} - -// Initialize receiver, resets codec database etc. -int32_t AudioCodingModuleImpl::InitializeReceiverSafe() { - initial_delay_ms_ = 0; - num_packets_accumulated_ = 0; - num_bytes_accumulated_ = 0; - accumulated_audio_ms_ = 0; - first_payload_received_ = 0; - last_incoming_send_timestamp_ = 0; - track_neteq_buffer_ = false; - playout_ts_ = 0; - // If the receiver is already initialized then we want to destroy any - // existing decoders. After a call to this function, we should have a clean - // start-up. - if (receiver_initialized_) { - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (UnregisterReceiveCodecSafe(i) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitializeReceiver() failed, Could not unregister codec"); - return -1; - } - } - } - if (neteq_.Init() != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "InitializeReceiver() failed, Could not initialize NetEQ"); - return -1; - } - neteq_.set_id(id_); - if (neteq_.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "NetEQ cannot allocate_packet Buffer"); - return -1; - } - - // Register RED and CN. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (IsCodecRED(i) || IsCodecCN(i)) { - if (RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i, - ACMNetEQ::kMasterJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - registered_pltypes_[i] = ACMCodecDB::database_[i].pltype; - } - } - - receiver_initialized_ = true; - return 0; -} - -// Reset the decoder state. -int32_t AudioCodingModuleImpl::ResetDecoder() { - CriticalSectionScoped lock(acm_crit_sect_); - - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if ((codecs_[id] != NULL) && (registered_pltypes_[id] != -1)) { - if (codecs_[id]->ResetDecoder(registered_pltypes_[id]) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "ResetDecoder failed:"); - return -1; - } - } - } - return neteq_.FlushBuffers(); -} - -// Get current receive frequency. -int32_t AudioCodingModuleImpl::ReceiveFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "ReceiveFrequency()"); - WebRtcACMCodecParams codec_params; - - CriticalSectionScoped lock(acm_crit_sect_); - if (DecoderParamByPlType(last_recv_audio_codec_pltype_, codec_params) < 0) { - return neteq_.CurrentSampFreqHz(); - } else if (codec_params.codec_inst.plfreq == 48000) { - // TODO(tlegrand): Remove this option when we have full 48 kHz support. - return 32000; - } else { - return codec_params.codec_inst.plfreq; - } -} - -// Get current playout frequency. -int32_t AudioCodingModuleImpl::PlayoutFrequency() const { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "PlayoutFrequency()"); - - CriticalSectionScoped lock(acm_crit_sect_); - - return neteq_.CurrentSampFreqHz(); -} - -// Register possible receive codecs, can be called multiple times, -// for codecs, CNG (NB, WB and SWB), DTMF, RED. -int32_t AudioCodingModuleImpl::RegisterReceiveCodec( - const CodecInst& receive_codec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (receive_codec.channels > 2) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "More than 2 audio channel is not supported."); - return -1; - } - - int mirror_id; - int codec_id = ACMCodecDB::ReceiverCodecNumber(&receive_codec, &mirror_id); - - if (codec_id < 0 || codec_id >= ACMCodecDB::kNumCodecs) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Wrong codec params to be registered as receive codec"); - return -1; - } - // Check if the payload-type is valid. - if (!ACMCodecDB::ValidPayloadType(receive_codec.pltype)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid payload-type %d for %s.", receive_codec.pltype, - receive_codec.plname); - return -1; - } - - if (!receiver_initialized_) { - if (InitializeReceiverSafe() < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot initialize reciver, so failed registering a codec."); - return -1; - } - } - - // If codec already registered, unregister. Except for CN where we only - // unregister if payload type is changing. - if ((registered_pltypes_[codec_id] == receive_codec.pltype) - && IsCodecCN(&receive_codec)) { - // Codec already registered as receiver with this payload type. Nothing - // to be done. - return 0; - } else if (registered_pltypes_[codec_id] != -1) { - if (UnregisterReceiveCodecSafe(codec_id) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - } - - if (RegisterRecCodecMSSafe(receive_codec, codec_id, mirror_id, - ACMNetEQ::kMasterJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register master codec."); - return -1; - } - - // TODO(andrew): Refactor how the slave is initialized. Can we instead - // always start up a slave and pre-register CN and RED? We should be able - // to get rid of stereo_receive_registered_. - // http://code.google.com/p/webrtc/issues/detail?id=453 - - // Register stereo codecs with the slave, or, if we've had already seen a - // stereo codec, register CN or RED as a special case. - if (receive_codec.channels == 2 || - (stereo_receive_registered_ && (IsCodecCN(&receive_codec) || - IsCodecRED(&receive_codec)))) { - // TODO(andrew): refactor this block to combine with InitStereoSlave(). - - if (!stereo_receive_registered_) { - // This is the first time a stereo codec has been registered. Make - // some stereo preparations. - - // Add a stereo slave. - assert(neteq_.num_slaves() == 0); - if (neteq_.AddSlave(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add slave jitter buffer to NetEQ."); - return -1; - } - - // Register any existing CN or RED codecs with the slave and as stereo. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (registered_pltypes_[i] != -1 && (IsCodecRED(i) || IsCodecCN(i))) { - stereo_receive_[i] = true; - - CodecInst codec; - memcpy(&codec, &ACMCodecDB::database_[i], sizeof(CodecInst)); - codec.pltype = registered_pltypes_[i]; - if (RegisterRecCodecMSSafe(codec, i, i, ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - } - } - } - - if (RegisterRecCodecMSSafe(receive_codec, codec_id, mirror_id, - ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - - if (!stereo_receive_[codec_id] && - (last_recv_audio_codec_pltype_ == receive_codec.pltype)) { - // The last received payload type is the same as the one we are - // registering. Expected number of channels to receive is one (mono), - // but we are now registering the receiving codec as stereo (number of - // channels is 2). - // Set |last_recv_audio_coded_pltype_| to invalid value to trigger a - // flush in NetEq, and a reset of expected number of channels next time a - // packet is received in AudioCodingModuleImpl::IncomingPacket(). - last_recv_audio_codec_pltype_ = -1; - } - - stereo_receive_[codec_id] = true; - stereo_receive_registered_ = true; - } else { - if (last_recv_audio_codec_pltype_ == receive_codec.pltype && - expected_channels_ == 2) { - // The last received payload type is the same as the one we are - // registering. Expected number of channels to receive is two (stereo), - // but we are now registering the receiving codec as mono (number of - // channels is 1). - // Set |last_recv_audio_coded_pl_type_| to invalid value to trigger a - // flush in NetEq, and a reset of expected number of channels next time a - // packet is received in AudioCodingModuleImpl::IncomingPacket(). - last_recv_audio_codec_pltype_ = -1; - } - stereo_receive_[codec_id] = false; - } - - registered_pltypes_[codec_id] = receive_codec.pltype; - - if (IsCodecRED(&receive_codec)) { - receive_red_pltype_ = receive_codec.pltype; - } - return 0; -} - -int32_t AudioCodingModuleImpl::RegisterRecCodecMSSafe( - const CodecInst& receive_codec, int16_t codec_id, - int16_t mirror_id, ACMNetEQ::JitterBuffer jitter_buffer) { - ACMGenericCodec** codecs; - if (jitter_buffer == ACMNetEQ::kMasterJb) { - codecs = &codecs_[0]; - } else if (jitter_buffer == ACMNetEQ::kSlaveJb) { - codecs = &slave_codecs_[0]; - if (codecs_[codec_id]->IsTrueStereoCodec()) { - // True stereo codecs need to use the same codec memory - // for both master and slave. - slave_codecs_[mirror_id] = codecs_[mirror_id]; - mirror_codec_idx_[mirror_id] = mirror_id; - } - } else { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "RegisterReceiveCodecMSSafe failed, jitter_buffer is neither " - "master or slave "); - return -1; - } - - if (codecs[mirror_id] == NULL) { - codecs[mirror_id] = CreateCodec(receive_codec); - if (codecs[mirror_id] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot create codec to register as receive codec"); - return -1; - } - mirror_codec_idx_[mirror_id] = mirror_id; - } - if (mirror_id != codec_id) { - codecs[codec_id] = codecs[mirror_id]; - mirror_codec_idx_[codec_id] = mirror_id; - } - - codecs[codec_id]->SetIsMaster(jitter_buffer == ACMNetEQ::kMasterJb); - - int16_t status = 0; - WebRtcACMCodecParams codec_params; - memcpy(&(codec_params.codec_inst), &receive_codec, sizeof(CodecInst)); - codec_params.enable_vad = false; - codec_params.enable_dtx = false; - codec_params.vad_mode = VADNormal; - if (!codecs[codec_id]->DecoderInitialized()) { - // Force initialization. - status = codecs[codec_id]->InitDecoder(&codec_params, true); - if (status < 0) { - // Could not initialize the decoder, we don't want to - // continue if we could not initialize properly. - WEBRTC_TRACE( - webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "could not initialize the receive codec, codec not registered"); - - return -1; - } - } else if (mirror_id != codec_id) { - // Currently this only happens for iSAC. - // We have to store the decoder parameters. - codecs[codec_id]->SaveDecoderParam(&codec_params); - } - - if (codecs[codec_id]->RegisterInNetEq(&neteq_, receive_codec) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Receive codec could not be registered in NetEQ"); - return -1; - } - // Guarantee that the same payload-type that is - // registered in NetEQ is stored in the codec. - codecs[codec_id]->SaveDecoderParam(&codec_params); - - return status; -} - -// Get current received codec. -int32_t AudioCodingModuleImpl::ReceiveCodec( - CodecInst* current_codec) const { - WebRtcACMCodecParams decoder_param; - CriticalSectionScoped lock(acm_crit_sect_); - - for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (codecs_[id] != NULL) { - if (codecs_[id]->DecoderInitialized()) { - if (codecs_[id]->DecoderParams(&decoder_param, - last_recv_audio_codec_pltype_)) { - memcpy(current_codec, &decoder_param.codec_inst, - sizeof(CodecInst)); - return 0; - } - } - } - } - - // If we are here then we haven't found any codec. Set codec pltype to -1 to - // indicate that the structure is invalid and return -1. - current_codec->pltype = -1; - return -1; -} - -// Incoming packet from network parsed and ready for decode. -int32_t AudioCodingModuleImpl::IncomingPacket( - const uint8_t* incoming_payload, - const int32_t payload_length, - const WebRtcRTPHeader& rtp_info) { - WebRtcRTPHeader rtp_header; - - memcpy(&rtp_header, &rtp_info, sizeof(WebRtcRTPHeader)); - - if (payload_length < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPacket() Error, payload-length cannot be negative"); - return -1; - } - - { - // Store the payload Type. This will be used to retrieve "received codec" - // and "received frequency." - CriticalSectionScoped lock(acm_crit_sect_); - - // Check there are packets missed between the last injected packet, and the - // latest received packet. If so and we are in AV-sync mode then we would - // like to fill the gap. Shouldn't be the first payload. - if (av_sync_ && first_payload_received_ && - rtp_info.header.sequenceNumber > last_sequence_number_ + 1) { - // If the last packet pushed was sync-packet account for all missing - // packets. Otherwise leave some room for PLC. - if (last_packet_was_sync_) { - while (rtp_info.header.sequenceNumber > last_sequence_number_ + 2) { - PushSyncPacketSafe(); - } - } else { - // Leave two packet room for NetEq perform PLC. - if (rtp_info.header.sequenceNumber > last_sequence_number_ + 3) { - last_sequence_number_ += 2; - last_incoming_send_timestamp_ += last_timestamp_diff_ * 2; - last_receive_timestamp_ += 2 * last_timestamp_diff_; - while (rtp_info.header.sequenceNumber > last_sequence_number_ + 1) - PushSyncPacketSafe(); - } - } - } - - uint8_t my_payload_type; - - // Check if this is an RED payload. - if (rtp_info.header.payloadType == receive_red_pltype_) { - // Get the primary payload-type. - my_payload_type = incoming_payload[0] & 0x7F; - } else { - my_payload_type = rtp_info.header.payloadType; - } - - // If payload is audio, check if received payload is different from - // previous. - if (!rtp_info.type.Audio.isCNG) { - // This is Audio not CNG. - - if (my_payload_type != last_recv_audio_codec_pltype_) { - // We detect a change in payload type. It is necessary for iSAC - // we are going to use ONE iSAC instance for decoding both WB and - // SWB payloads. If payload is changed there might be a need to reset - // sampling rate of decoder. depending what we have received "now". - for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) { - if (registered_pltypes_[i] == my_payload_type) { - if (UpdateUponReceivingCodec(i) != 0) - return -1; - break; - } - } - // Codec is changed, there might be a jump in timestamp, therefore, - // we have to reset some variables that track NetEq buffer. - if (track_neteq_buffer_ || av_sync_) { - last_incoming_send_timestamp_ = rtp_info.header.timestamp; - } - - if (nack_enabled_) { - assert(nack_.get()); - // Codec is changed, reset NACK and update sampling rate. - nack_->Reset(); - nack_->UpdateSampleRate( - ACMCodecDB::database_[current_receive_codec_idx_].plfreq); - } - } - last_recv_audio_codec_pltype_ = my_payload_type; - } - - // Current timestamp based on the receiver sampling frequency. - last_receive_timestamp_ = NowTimestamp(current_receive_codec_idx_); - - if (nack_enabled_) { - assert(nack_.get()); - nack_->UpdateLastReceivedPacket(rtp_header.header.sequenceNumber, - rtp_header.header.timestamp); - } - } - - int per_neteq_payload_length = payload_length; - // Split the payload for stereo packets, so that first half of payload - // vector holds left channel, and second half holds right channel. - if (expected_channels_ == 2) { - if (!rtp_info.type.Audio.isCNG) { - // Create a new vector for the payload, maximum payload size. - int32_t length = payload_length; - uint8_t payload[kMaxPacketSize]; - assert(payload_length <= kMaxPacketSize); - memcpy(payload, incoming_payload, payload_length); - codecs_[current_receive_codec_idx_]->SplitStereoPacket(payload, &length); - rtp_header.type.Audio.channel = 2; - per_neteq_payload_length = length / 2; - // Insert packet into NetEQ. - if (neteq_.RecIn(payload, length, rtp_header, - last_receive_timestamp_) < 0) - return -1; - } else { - // If we receive a CNG packet while expecting stereo, we ignore the - // packet and continue. CNG is not supported for stereo. - return 0; - } - } else { - if (neteq_.RecIn(incoming_payload, payload_length, rtp_header, - last_receive_timestamp_) < 0) - return -1; - } - - { - CriticalSectionScoped lock(acm_crit_sect_); - - // Update buffering uses |last_incoming_send_timestamp_| so it should be - // before the next block. - if (track_neteq_buffer_) - UpdateBufferingSafe(rtp_header, per_neteq_payload_length); - - if (av_sync_) { - if (rtp_info.header.sequenceNumber == last_sequence_number_ + 1) { - last_timestamp_diff_ = rtp_info.header.timestamp - - last_incoming_send_timestamp_; - } - last_sequence_number_ = rtp_info.header.sequenceNumber; - last_ssrc_ = rtp_info.header.ssrc; - last_packet_was_sync_ = false; - } - - if (av_sync_ || track_neteq_buffer_) { - last_incoming_send_timestamp_ = rtp_info.header.timestamp; - } - - // Set the following regardless of tracking NetEq buffer or being in - // AV-sync mode. Only if the received packet is not CNG. - if (!rtp_info.type.Audio.isCNG) - first_payload_received_ = true; - } - return 0; -} - -int AudioCodingModuleImpl::UpdateUponReceivingCodec(int index) { - if (codecs_[index] == NULL) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "IncomingPacket() error: payload type found but " - "corresponding codec is NULL"); - return -1; - } - codecs_[index]->UpdateDecoderSampFreq(index); - neteq_.set_received_stereo(stereo_receive_[index]); - current_receive_codec_idx_ = index; - - // If we have a change in the expected number of channels, flush packet - // buffers in NetEQ. - if ((stereo_receive_[index] && (expected_channels_ == 1)) || - (!stereo_receive_[index] && (expected_channels_ == 2))) { - neteq_.FlushBuffers(); - codecs_[index]->ResetDecoder(registered_pltypes_[index]); - } - - if (stereo_receive_[index] && (expected_channels_ == 1)) { - // When switching from a mono to stereo codec reset the slave. - if (InitStereoSlave() != 0) - return -1; - } - - // Store number of channels we expect to receive for the current payload type. - if (stereo_receive_[index]) { - expected_channels_ = 2; - } else { - expected_channels_ = 1; - } - - // Reset previous received channel. - prev_received_channel_ = 0; - return 0; -} - -bool AudioCodingModuleImpl::IsCodecForSlave(int index) const { - return (registered_pltypes_[index] != -1 && stereo_receive_[index]); -} - -int AudioCodingModuleImpl::InitStereoSlave() { - neteq_.RemoveSlaves(); - - if (neteq_.AddSlave(ACMCodecDB::NetEQDecoders(), - ACMCodecDB::kNumCodecs) < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot add slave jitter buffer to NetEQ."); - return -1; - } - - // Register all needed codecs with slave. - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (codecs_[i] != NULL && IsCodecForSlave(i)) { - WebRtcACMCodecParams decoder_params; - if (codecs_[i]->DecoderParams(&decoder_params, registered_pltypes_[i])) { - if (RegisterRecCodecMSSafe(decoder_params.codec_inst, - i, ACMCodecDB::MirrorID(i), - ACMNetEQ::kSlaveJb) < 0) { - WEBRTC_TRACE(kTraceError, kTraceAudioCoding, id_, - "Cannot register slave codec."); - return -1; - } - } - } - } - return 0; -} - -int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) { - { - CriticalSectionScoped lock(acm_crit_sect_); - // Don't let the extra delay modified while accumulating buffers in NetEq. - if (track_neteq_buffer_ && first_payload_received_) - return 0; - } - return neteq_.SetMinimumDelay(time_ms); -} - -int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) { - return neteq_.SetMaximumDelay(time_ms); -} - -// Get Dtmf playout status. -bool AudioCodingModuleImpl::DtmfPlayoutStatus() const { -#ifndef WEBRTC_CODEC_AVT - return false; -#else - return neteq_.avt_playout(); -#endif -} - -// Configure Dtmf playout status i.e on/off playout the incoming outband -// Dtmf tone. -int32_t AudioCodingModuleImpl::SetDtmfPlayoutStatus( -#ifndef WEBRTC_CODEC_AVT - const bool /* enable */) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "SetDtmfPlayoutStatus() failed: AVT is not supported."); - return -1; -#else - const bool enable) { - return neteq_.SetAVTPlayout(enable); -#endif -} - -// Estimate the Bandwidth based on the incoming stream, needed for one way -// audio where the RTCP send the BW estimate. -// This is also done in the RTP module. -int32_t AudioCodingModuleImpl::DecoderEstimatedBandwidth() const { - CodecInst codec; - int16_t codec_id = -1; - int pltype_wb; - int pltype_swb; - - // Get iSAC settings. - for (int id = 0; id < ACMCodecDB::kNumCodecs; id++) { - // Store codec settings for codec number "codeCntr" in the output struct. - ACMCodecDB::Codec(id, &codec); - - if (!STR_CASE_CMP(codec.plname, "isac")) { - codec_id = 1; - pltype_wb = codec.pltype; - - ACMCodecDB::Codec(id + 1, &codec); - pltype_swb = codec.pltype; - - break; - } - } - - if (codec_id < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "DecoderEstimatedBandwidth failed"); - return -1; - } - - if ((last_recv_audio_codec_pltype_ == pltype_wb) || - (last_recv_audio_codec_pltype_ == pltype_swb)) { - return codecs_[codec_id]->GetEstimatedBandwidth(); - } else { - return -1; - } -} - -// Set playout mode for: voice, fax, or streaming. -int32_t AudioCodingModuleImpl::SetPlayoutMode( - const AudioPlayoutMode mode) { - if ((mode != voice) && (mode != fax) && (mode != streaming) && - (mode != off)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Invalid playout mode."); - return -1; - } - return neteq_.SetPlayoutMode(mode); -} - -// Get playout mode voice, fax. -AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const { - return neteq_.playout_mode(); -} - -// Get 10 milliseconds of raw audio data to play out. -// Automatic resample to the requested frequency. -int32_t AudioCodingModuleImpl::PlayoutData10Ms( - int32_t desired_freq_hz, AudioFrame* audio_frame) { - TRACE_EVENT_ASYNC_BEGIN0("webrtc", "ACM::PlayoutData10Ms", this); - bool stereo_mode; - - if (GetSilence(desired_freq_hz, audio_frame)) { - TRACE_EVENT_ASYNC_END1("webrtc", "ACM::PlayoutData10Ms", this, - "silence", true); - return 0; // Silence is generated, return. - } - - // RecOut always returns 10 ms. - if (neteq_.RecOut(audio_frame_) != 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "PlayoutData failed, RecOut Failed"); - return -1; - } - int decoded_seq_num; - uint32_t decoded_timestamp; - bool update_nack = - neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp) && - nack_enabled_; // Update NACK only if it is enabled. - audio_frame->num_channels_ = audio_frame_.num_channels_; - audio_frame->vad_activity_ = audio_frame_.vad_activity_; - audio_frame->speech_type_ = audio_frame_.speech_type_; - - stereo_mode = (audio_frame_.num_channels_ > 1); - - // For stereo playout: - // Master and Slave samples are interleaved starting with Master. - const uint16_t receive_freq = - static_cast<uint16_t>(audio_frame_.sample_rate_hz_); - bool tone_detected = false; - int16_t last_detected_tone; - int16_t tone; - - // Limit the scope of ACM Critical section. - { - CriticalSectionScoped lock(acm_crit_sect_); - - // Update call statistics. - call_stats_.DecodedByNetEq(audio_frame->speech_type_); - - if (update_nack) { - assert(nack_.get()); - nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp); - } - - // If we are in AV-sync and have already received an audio packet, but the - // latest packet is too late, then insert sync packet. - if (av_sync_ && first_payload_received_ && - NowTimestamp(current_receive_codec_idx_) > 5 * last_timestamp_diff_ + - last_receive_timestamp_) { - if (!last_packet_was_sync_) { - // If the last packet inserted has been a regular packet Skip two - // packets to give room for PLC. - last_incoming_send_timestamp_ += 2 * last_timestamp_diff_; - last_sequence_number_ += 2; - last_receive_timestamp_ += 2 * last_timestamp_diff_; - } - - // One sync packet. - if (PushSyncPacketSafe() < 0) - return -1; - } - - if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) { - TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", this, - "seqnum", decoded_seq_num, - "now", clock_->TimeInMilliseconds()); - // Resample payload_data. - int16_t temp_len = output_resampler_.Resample10Msec( - audio_frame_.data_, receive_freq, audio_frame->data_, - desired_freq_hz, audio_frame_.num_channels_); - - if (temp_len < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "PlayoutData failed, resampler failed"); - return -1; - } - - // Set the payload data length from the resampler. - audio_frame->samples_per_channel_ = static_cast<uint16_t>(temp_len); - // Set the sampling frequency. - audio_frame->sample_rate_hz_ = desired_freq_hz; - } else { - TRACE_EVENT_ASYNC_END2("webrtc", "ACM::PlayoutData10Ms", this, - "seqnum", decoded_seq_num, - "now", clock_->TimeInMilliseconds()); - memcpy(audio_frame->data_, audio_frame_.data_, - audio_frame_.samples_per_channel_ * audio_frame->num_channels_ - * sizeof(int16_t)); - // Set the payload length. - audio_frame->samples_per_channel_ = - audio_frame_.samples_per_channel_; - // Set the sampling frequency. - audio_frame->sample_rate_hz_ = receive_freq; - } - - // Tone detection done for master channel. - if (dtmf_detector_ != NULL) { - // Dtmf Detection. - if (audio_frame->sample_rate_hz_ == 8000) { - // Use audio_frame->data_ then Dtmf detector doesn't - // need resampling. - if (!stereo_mode) { - dtmf_detector_->Detect(audio_frame->data_, - audio_frame->samples_per_channel_, - audio_frame->sample_rate_hz_, tone_detected, - tone); - } else { - // We are in 8 kHz so the master channel needs only 80 samples. - int16_t master_channel[80]; - for (int n = 0; n < 80; n++) { - master_channel[n] = audio_frame->data_[n << 1]; - } - dtmf_detector_->Detect(master_channel, - audio_frame->samples_per_channel_, - audio_frame->sample_rate_hz_, tone_detected, - tone); - } - } else { - // Do the detection on the audio that we got from NetEQ (audio_frame_). - if (!stereo_mode) { - dtmf_detector_->Detect(audio_frame_.data_, - audio_frame_.samples_per_channel_, - receive_freq, tone_detected, tone); - } else { - int16_t master_channel[WEBRTC_10MS_PCM_AUDIO]; - for (int n = 0; n < audio_frame_.samples_per_channel_; n++) { - master_channel[n] = audio_frame_.data_[n << 1]; - } - dtmf_detector_->Detect(master_channel, - audio_frame_.samples_per_channel_, - receive_freq, tone_detected, tone); - } - } - } - - // We want to do this while we are in acm_crit_sect_. - // (Doesn't really need to initialize the following - // variable but Linux complains if we don't.) - last_detected_tone = kACMToneEnd; - if (tone_detected) { - last_detected_tone = last_detected_tone_; - last_detected_tone_ = tone; - } - } - - if (tone_detected) { - // We will deal with callback here, so enter callback critical section. - CriticalSectionScoped lock(callback_crit_sect_); - - if (dtmf_callback_ != NULL) { - if (tone != kACMToneEnd) { - // just a tone - dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(tone), false); - } else if ((tone == kACMToneEnd) && (last_detected_tone != kACMToneEnd)) { - // The tone is "END" and the previously detected tone is - // not "END," so call fir an end. - dtmf_callback_->IncomingDtmf(static_cast<uint8_t>(last_detected_tone), - true); - } - } - } - - audio_frame->id_ = id_; - audio_frame->energy_ = -1; - audio_frame->timestamp_ = 0; - - return 0; -} - -///////////////////////////////////////// -// Statistics -// - -int32_t AudioCodingModuleImpl::NetworkStatistics( - ACMNetworkStatistics* statistics) { - int32_t status; - status = neteq_.NetworkStatistics(statistics); - return status; -} - -void AudioCodingModuleImpl::DestructEncoderInst(void* inst) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "DestructEncoderInst()"); - if (!HaveValidEncoder("DestructEncoderInst")) { - return; - } - - codecs_[current_send_codec_idx_]->DestructEncoderInst(inst); -} - -int16_t AudioCodingModuleImpl::AudioBuffer( - WebRtcACMAudioBuff& buffer) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "AudioBuffer()"); - if (!HaveValidEncoder("AudioBuffer")) { - return -1; - } - buffer.last_in_timestamp = last_in_timestamp_; - return codecs_[current_send_codec_idx_]->AudioBuffer(buffer); -} - -int16_t AudioCodingModuleImpl::SetAudioBuffer( - WebRtcACMAudioBuff& buffer) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "SetAudioBuffer()"); - if (!HaveValidEncoder("SetAudioBuffer")) { - return -1; - } - return codecs_[current_send_codec_idx_]->SetAudioBuffer(buffer); -} - -uint32_t AudioCodingModuleImpl::EarliestTimestamp() const { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "EarliestTimestamp()"); - if (!HaveValidEncoder("EarliestTimestamp")) { - return -1; - } - return codecs_[current_send_codec_idx_]->EarliestTimestamp(); -} - -int32_t AudioCodingModuleImpl::RegisterVADCallback( - ACMVADCallback* vad_callback) { - WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, - "RegisterVADCallback()"); - CriticalSectionScoped lock(callback_crit_sect_); - vad_callback_ = vad_callback; - return 0; -} - -// TODO(turajs): Remove this API if it is not used. -// TODO(tlegrand): Modify this function to work for stereo, and add tests. -// TODO(turajs): Receive timestamp in this method is incremented by frame-size -// and does not reflect the true receive frame-size. Therefore, subsequent -// jitter computations are not accurate. -int32_t AudioCodingModuleImpl::IncomingPayload( - const uint8_t* incoming_payload, const int32_t payload_length, - const uint8_t payload_type, const uint32_t timestamp) { - if (payload_length < 0) { - // Log error in trace file. - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPacket() Error, payload-length cannot be negative"); - return -1; - } - - if (dummy_rtp_header_ == NULL) { - // This is the first time that we are using |dummy_rtp_header_| - // so we have to create it. - WebRtcACMCodecParams codec_params; - dummy_rtp_header_ = new WebRtcRTPHeader; - if (dummy_rtp_header_ == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "IncomingPayload() Error, out of memory"); - return -1; - } - dummy_rtp_header_->header.payloadType = payload_type; - // Don't matter in this case. - dummy_rtp_header_->header.ssrc = 0; - dummy_rtp_header_->header.markerBit = false; - // Start with random numbers. - dummy_rtp_header_->header.sequenceNumber = rand(); - dummy_rtp_header_->header.timestamp = - (static_cast<uint32_t>(rand()) << 16) + - static_cast<uint32_t>(rand()); - dummy_rtp_header_->type.Audio.channel = 1; - - if (DecoderParamByPlType(payload_type, codec_params) < 0) { - // We didn't find a codec with the given payload. - // Something is wrong we exit, but we delete |dummy_rtp_header_| - // and set it to NULL to start clean next time. - delete dummy_rtp_header_; - dummy_rtp_header_ = NULL; - return -1; - } - recv_pl_frame_size_smpls_ = codec_params.codec_inst.pacsize; - } - - if (payload_type != dummy_rtp_header_->header.payloadType) { - // Payload type has changed since the last time we might need to - // update the frame-size. - WebRtcACMCodecParams codec_params; - if (DecoderParamByPlType(payload_type, codec_params) < 0) { - // We didn't find a codec with the given payload. - return -1; - } - recv_pl_frame_size_smpls_ = codec_params.codec_inst.pacsize; - dummy_rtp_header_->header.payloadType = payload_type; - } - - if (timestamp > 0) { - dummy_rtp_header_->header.timestamp = timestamp; - } - - // Store the payload Type. this will be used to retrieve "received codec" - // and "received frequency." - last_recv_audio_codec_pltype_ = payload_type; - - last_receive_timestamp_ += recv_pl_frame_size_smpls_; - // Insert in NetEQ. - if (neteq_.RecIn(incoming_payload, payload_length, *dummy_rtp_header_, - last_receive_timestamp_) < 0) { - return -1; - } - - // Get ready for the next payload. - dummy_rtp_header_->header.sequenceNumber++; - dummy_rtp_header_->header.timestamp += recv_pl_frame_size_smpls_; - return 0; -} - -int16_t AudioCodingModuleImpl::DecoderParamByPlType( - const uint8_t payload_type, - WebRtcACMCodecParams& codec_params) const { - CriticalSectionScoped lock(acm_crit_sect_); - for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs; - id++) { - if (codecs_[id] != NULL) { - if (codecs_[id]->DecoderInitialized()) { - if (codecs_[id]->DecoderParams(&codec_params, payload_type)) { - return 0; - } - } - } - } - // If we are here it means that we could not find a - // codec with that payload type. reset the values to - // not acceptable values and return -1. - codec_params.codec_inst.plname[0] = '\0'; - codec_params.codec_inst.pacsize = 0; - codec_params.codec_inst.rate = 0; - codec_params.codec_inst.pltype = -1; - return -1; -} - -int16_t AudioCodingModuleImpl::DecoderListIDByPlName( - const char* name, const uint16_t frequency) const { - WebRtcACMCodecParams codec_params; - CriticalSectionScoped lock(acm_crit_sect_); - for (int16_t id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if ((codecs_[id] != NULL)) { - if (codecs_[id]->DecoderInitialized()) { - assert(registered_pltypes_[id] >= 0); - assert(registered_pltypes_[id] <= 255); - codecs_[id]->DecoderParams( - &codec_params, static_cast<uint8_t>(registered_pltypes_[id])); - if (!STR_CASE_CMP(codec_params.codec_inst.plname, name)) { - // Check if the given sampling frequency matches. - // A zero sampling frequency means we matching the names - // is sufficient and we don't need to check for the - // frequencies. - // Currently it is only iSAC which has one name but two - // sampling frequencies. - if ((frequency == 0)|| - (codec_params.codec_inst.plfreq == frequency)) { - return id; - } - } - } - } - } - // If we are here it means that we could not find a - // codec with that payload type. return -1. - return -1; -} - -int32_t AudioCodingModuleImpl::LastEncodedTimestamp( - uint32_t& timestamp) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (!HaveValidEncoder("LastEncodedTimestamp")) { - return -1; - } - timestamp = codecs_[current_send_codec_idx_]->LastEncodedTimestamp(); - return 0; -} - -int32_t AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc( - bool use_webrtc_dtx) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) { - WEBRTC_TRACE( - webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Cannot replace codec internal DTX when no send codec is registered."); - return -1; - } - - int32_t res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX( - use_webrtc_dtx); - // Check if VAD is turned on, or if there is any error. - if (res == 1) { - vad_enabled_ = true; - } else if (res < 0) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Failed to set ReplaceInternalDTXWithWebRtc(%d)", - use_webrtc_dtx); - return res; - } - - return 0; -} - -int32_t AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc( - bool* uses_webrtc_dtx) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) { - return -1; - } - if (codecs_[current_send_codec_idx_]->IsInternalDTXReplaced(uses_webrtc_dtx) - < 0) { - return -1; - } - return 0; -} - -int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("SetISACMaxRate")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->SetISACMaxRate(max_bit_per_sec); -} - -int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("SetISACMaxPayloadSize")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->SetISACMaxPayloadSize( - max_size_bytes); -} - -int32_t AudioCodingModuleImpl::ConfigISACBandwidthEstimator( - int frame_size_ms, - int rate_bit_per_sec, - bool enforce_frame_size) { - CriticalSectionScoped lock(acm_crit_sect_); - - if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) { - return -1; - } - - return codecs_[current_send_codec_idx_]->ConfigISACBandwidthEstimator( - frame_size_ms, rate_bit_per_sec, enforce_frame_size); -} - -int32_t AudioCodingModuleImpl::PlayoutTimestamp( - uint32_t* timestamp) { - WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, - "PlayoutTimestamp()"); - { - CriticalSectionScoped lock(acm_crit_sect_); - if (track_neteq_buffer_) { - *timestamp = playout_ts_; - return 0; - } - } - return neteq_.PlayoutTimestamp(*timestamp); -} - -bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { - if ((!send_codec_registered_) || (current_send_codec_idx_ < 0) || - (current_send_codec_idx_ >= ACMCodecDB::kNumCodecs)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: No send codec is registered.", caller_name); - return false; - } - if ((current_send_codec_idx_ < 0) || - (current_send_codec_idx_ >= ACMCodecDB::kNumCodecs)) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: Send codec index out of range.", caller_name); - return false; - } - if (codecs_[current_send_codec_idx_] == NULL) { - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "%s failed: Send codec is NULL pointer.", caller_name); - return false; - } - return true; -} - -int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { - CriticalSectionScoped lock(acm_crit_sect_); - int id; - - // Search through the list of registered payload types. - for (id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) { - if (registered_pltypes_[id] == payload_type) { - // We have found the id registered with the payload type. - break; - } - } - - if (id >= ACMCodecDB::kNumCodecs) { - // Payload type was not registered. No need to unregister. - return 0; - } - - // Unregister the codec with the given payload type. - return UnregisterReceiveCodecSafe(id); -} - -int32_t AudioCodingModuleImpl::UnregisterReceiveCodecSafe( - const int16_t codec_id) { - const WebRtcNetEQDecoder *neteq_decoder = ACMCodecDB::NetEQDecoders(); - int16_t mirror_id = ACMCodecDB::MirrorID(codec_id); - bool stereo_receiver = false; - - if (codecs_[codec_id] != NULL) { - if (registered_pltypes_[codec_id] != -1) { - // Store stereo information for future use. - stereo_receiver = stereo_receive_[codec_id]; - - // Before deleting the decoder instance unregister from NetEQ. - if (neteq_.RemoveCodec(neteq_decoder[codec_id], - stereo_receive_[codec_id]) < 0) { - CodecInst codec; - ACMCodecDB::Codec(codec_id, &codec); - WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, - "Unregistering %s-%d from NetEQ failed.", codec.plname, - codec.plfreq); - return -1; - } - - // CN is a special case for NetEQ, all three sampling frequencies - // are unregistered if one is deleted. - if (IsCodecCN(codec_id)) { - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (IsCodecCN(i)) { - stereo_receive_[i] = false; - registered_pltypes_[i] = -1; - } - } - } else { - if (codec_id == mirror_id) { - codecs_[codec_id]->DestructDecoder(); - if (stereo_receive_[codec_id]) { - slave_codecs_[codec_id]->DestructDecoder(); - stereo_receive_[codec_id] = false; - } - } - } - - // Check if this is the last registered stereo receive codec. - if (stereo_receiver) { - bool no_stereo = true; - - for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { - if (stereo_receive_[i]) { - // We still have stereo codecs registered. - no_stereo = false; - break; - } - } - - // If we don't have any stereo codecs left, change status. - if (no_stereo) { - neteq_.RemoveSlaves(); // No longer need the slave. - stereo_receive_registered_ = false; - } - } - } - } - - if (registered_pltypes_[codec_id] == receive_red_pltype_) { - // RED is going to be unregistered, set to an invalid value. - receive_red_pltype_ = 255; - } - registered_pltypes_[codec_id] = -1; - - return 0; -} - -int32_t AudioCodingModuleImpl::REDPayloadISAC( - const int32_t isac_rate, const int16_t isac_bw_estimate, - uint8_t* payload, int16_t* length_bytes) { - if (!HaveValidEncoder("EncodeData")) { - return -1; - } - int16_t status; - status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate, - isac_bw_estimate, - payload, - length_bytes); - return status; -} - -void AudioCodingModuleImpl::ResetFragmentation(int vector_size) { - for (int n = 0; n < kMaxNumFragmentationVectors; n++) { - fragmentation_.fragmentationOffset[n] = n * MAX_PAYLOAD_SIZE_BYTE; - } - memset(fragmentation_.fragmentationLength, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationLength[0])); - memset(fragmentation_.fragmentationTimeDiff, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationTimeDiff[0])); - memset(fragmentation_.fragmentationPlType, 0, kMaxNumFragmentationVectors * - sizeof(fragmentation_.fragmentationPlType[0])); - fragmentation_.fragmentationVectorSize = - static_cast<uint16_t>(vector_size); -} - -// TODO(turajs): Add second parameter to enable/disable AV-sync. -int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) { - if (delay_ms < 0 || delay_ms > 10000) { - return -1; - } - - CriticalSectionScoped lock(acm_crit_sect_); - - // Receiver should be initialized before this call processed. - if (!receiver_initialized_) { - InitializeReceiverSafe(); - } - - if (first_payload_received_) { - // Too late for this API. Only works before a call is started. - return -1; - } - initial_delay_ms_ = delay_ms; - - // If initial delay is zero, NetEq buffer should not be tracked, also we - // don't want to be in AV-sync mode. - track_neteq_buffer_ = delay_ms > 0; - av_sync_ = delay_ms > 0; - - neteq_.EnableAVSync(av_sync_); - return neteq_.SetMinimumDelay(delay_ms); -} - -bool AudioCodingModuleImpl::GetSilence(int desired_sample_rate_hz, - AudioFrame* frame) { - CriticalSectionScoped lock(acm_crit_sect_); - if (initial_delay_ms_ == 0 || !track_neteq_buffer_) { - return false; - } - - if (accumulated_audio_ms_ >= initial_delay_ms_) { - // We have enough data stored that match our initial delay target. - track_neteq_buffer_ = false; - return false; - } - - // Record call to silence generator. - call_stats_.DecodedBySilenceGenerator(); - - // We stop accumulating packets, if the number of packets or the total size - // exceeds a threshold. - int max_num_packets; - int buffer_size_bytes; - int per_payload_overhead_bytes; - neteq_.BufferSpec(max_num_packets, buffer_size_bytes, - per_payload_overhead_bytes); - int total_bytes_accumulated = num_bytes_accumulated_ + - num_packets_accumulated_ * per_payload_overhead_bytes; - if (num_packets_accumulated_ > max_num_packets * 0.9 || - total_bytes_accumulated > buffer_size_bytes * 0.9) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "GetSilence: Initial delay couldn't be achieved." - " num_packets_accumulated=%d, total_bytes_accumulated=%d", - num_packets_accumulated_, num_bytes_accumulated_); - track_neteq_buffer_ = false; - return false; - } - - if (desired_sample_rate_hz > 0) { - frame->sample_rate_hz_ = desired_sample_rate_hz; - } else { - frame->sample_rate_hz_ = 0; - if (current_receive_codec_idx_ >= 0) { - frame->sample_rate_hz_ = - ACMCodecDB::database_[current_receive_codec_idx_].plfreq; - } else { - // No payload received yet, use the default sampling rate of NetEq. - frame->sample_rate_hz_ = neteq_.CurrentSampFreqHz(); - } - } - frame->num_channels_ = expected_channels_; - frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms. - frame->speech_type_ = AudioFrame::kCNG; - frame->vad_activity_ = AudioFrame::kVadPassive; - frame->energy_ = 0; - int samples = frame->samples_per_channel_ * frame->num_channels_; - memset(frame->data_, 0, samples * sizeof(int16_t)); - return true; -} - -// Must be called within the scope of ACM critical section. -int AudioCodingModuleImpl::PushSyncPacketSafe() { - assert(av_sync_); - last_sequence_number_++; - last_incoming_send_timestamp_ += last_timestamp_diff_; - last_receive_timestamp_ += last_timestamp_diff_; - - WebRtcRTPHeader rtp_info; - rtp_info.header.payloadType = last_recv_audio_codec_pltype_; - rtp_info.header.ssrc = last_ssrc_; - rtp_info.header.markerBit = false; - rtp_info.header.sequenceNumber = last_sequence_number_; - rtp_info.header.timestamp = last_incoming_send_timestamp_; - rtp_info.type.Audio.channel = stereo_receive_[current_receive_codec_idx_] ? - 2 : 1; - last_packet_was_sync_ = true; - int payload_len_bytes = neteq_.RecIn(rtp_info, last_receive_timestamp_); - - if (payload_len_bytes < 0) - return -1; - - // This is to account for sync packets inserted during the buffering phase. - if (track_neteq_buffer_) - UpdateBufferingSafe(rtp_info, payload_len_bytes); - - return 0; -} - -// Must be called within the scope of ACM critical section. -void AudioCodingModuleImpl::UpdateBufferingSafe(const WebRtcRTPHeader& rtp_info, - int payload_len_bytes) { - const int in_sample_rate_khz = - (ACMCodecDB::database_[current_receive_codec_idx_].plfreq / 1000); - if (first_payload_received_ && - rtp_info.header.timestamp > last_incoming_send_timestamp_ && - in_sample_rate_khz > 0) { - accumulated_audio_ms_ += (rtp_info.header.timestamp - - last_incoming_send_timestamp_) / in_sample_rate_khz; - } - - num_packets_accumulated_++; - num_bytes_accumulated_ += payload_len_bytes; - - playout_ts_ = static_cast<uint32_t>( - rtp_info.header.timestamp - static_cast<uint32_t>( - initial_delay_ms_ * in_sample_rate_khz)); -} - -uint32_t AudioCodingModuleImpl::NowTimestamp(int codec_id) { - // Down-cast the time to (32-6)-bit since we only care about - // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms. - // we masked 6 most significant bits of 32-bit so we don't lose resolution - // when do the following multiplication. - int sample_rate_khz = ACMCodecDB::database_[codec_id].plfreq / 1000; - const uint32_t now_in_ms = static_cast<uint32_t>( - clock_->TimeInMilliseconds() & kMaskTimestamp); - return static_cast<uint32_t>(sample_rate_khz * now_in_ms); -} - -std::vector<uint16_t> AudioCodingModuleImpl::GetNackList( - int round_trip_time_ms) const { - CriticalSectionScoped lock(acm_crit_sect_); - if (round_trip_time_ms < 0) { - WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, - "GetNackList: round trip time cannot be negative." - " round_trip_time_ms=%d", round_trip_time_ms); - } - if (nack_enabled_ && round_trip_time_ms >= 0) { - assert(nack_.get()); - return nack_->GetNackList(round_trip_time_ms); - } - std::vector<uint16_t> empty_list; - return empty_list; -} - -int AudioCodingModuleImpl::LeastRequiredDelayMs() const { - return std::max(neteq_.LeastRequiredDelayMs(), initial_delay_ms_); -} - -int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) { - // Don't do anything if |max_nack_list_size| is out of range. - if (max_nack_list_size == 0 || - max_nack_list_size > acm2::Nack::kNackListSizeLimit) - return -1; - - CriticalSectionScoped lock(acm_crit_sect_); - if (!nack_enabled_) { - nack_.reset(acm2::Nack::Create(kNackThresholdPackets)); - nack_enabled_ = true; - - // Sampling rate might need to be updated if we change from disable to - // enable. Do it if the receive codec is valid. - if (current_receive_codec_idx_ >= 0) { - nack_->UpdateSampleRate( - ACMCodecDB::database_[current_receive_codec_idx_].plfreq); - } - } - return nack_->SetMaxNackListSize(max_nack_list_size); -} - -void AudioCodingModuleImpl::DisableNack() { - CriticalSectionScoped lock(acm_crit_sect_); - nack_.reset(); // Memory is released. - nack_enabled_ = false; -} - -const char* AudioCodingModuleImpl::Version() const { - return kLegacyAcmVersion; -} - -void AudioCodingModuleImpl::GetDecodingCallStatistics( - AudioDecodingCallStats* call_stats) const { - CriticalSectionScoped lock(acm_crit_sect_); - *call_stats = call_stats_.GetDecodingStatistics(); -} - -} // namespace acm1 - -} // namespace webrtc |