/* * 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/acm_g7291.h" #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h" #include "webrtc/modules/audio_coding/main/source/acm_neteq.h" #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" #include "webrtc/system_wrappers/interface/trace.h" #ifdef WEBRTC_CODEC_G729_1 // NOTE! G.729.1 is not included in the open-source package. Modify this file // or your codec API to match the function calls and names of used G.729.1 API // file. #include "g7291_interface.h" #endif namespace webrtc { #ifndef WEBRTC_CODEC_G729_1 ACMG729_1::ACMG729_1(int16_t /* codec_id */) : encoder_inst_ptr_(NULL), decoder_inst_ptr_(NULL), my_rate_(32000), flag_8khz_(0), flag_g729_mode_(0) { return; } ACMG729_1::~ACMG729_1() { return; } int16_t ACMG729_1::InternalEncode( uint8_t* /* bitstream */, int16_t* /* bitstream_len_byte */) { return -1; } int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */, int16_t /* bitstream_len_byte */, int16_t* /* audio */, int16_t* /* audio_samples */, int8_t* /* speech_type */) { return -1; } int16_t ACMG729_1::InternalInitEncoder( WebRtcACMCodecParams* /* codec_params */) { return -1; } int16_t ACMG729_1::InternalInitDecoder( WebRtcACMCodecParams* /* codec_params */) { return -1; } int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */, const CodecInst& /* codec_inst */) { return -1; } ACMGenericCodec* ACMG729_1::CreateInstance(void) { return NULL; } int16_t ACMG729_1::InternalCreateEncoder() { return -1; } void ACMG729_1::DestructEncoderSafe() { return; } int16_t ACMG729_1::InternalCreateDecoder() { return -1; } void ACMG729_1::DestructDecoderSafe() { return; } void ACMG729_1::InternalDestructEncoderInst(void* /* ptr_inst */) { return; } int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) { return -1; } #else //===================== Actual Implementation ======================= struct G729_1_inst_t_; ACMG729_1::ACMG729_1(int16_t codec_id) : encoder_inst_ptr_(NULL), decoder_inst_ptr_(NULL), my_rate_(32000), // Default rate. flag_8khz_(0), flag_g729_mode_(0) { // TODO(tlegrand): We should add codec_id as a input variable to the // constructor of ACMGenericCodec. codec_id_ = codec_id; return; } ACMG729_1::~ACMG729_1() { if (encoder_inst_ptr_ != NULL) { WebRtcG7291_Free(encoder_inst_ptr_); encoder_inst_ptr_ = NULL; } if (decoder_inst_ptr_ != NULL) { WebRtcG7291_Free(decoder_inst_ptr_); decoder_inst_ptr_ = NULL; } return; } int16_t ACMG729_1::InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte) { // Initialize before entering the loop int16_t num_encoded_samples = 0; *bitstream_len_byte = 0; int16_t byte_length_frame = 0; // Derive number of 20ms frames per encoded packet. // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples int16_t num_20ms_frames = (frame_len_smpl_ / 320); // Byte length for the frame. +1 is for rate information. byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 - flag_g729_mode_); // The following might be revised if we have G729.1 Annex C (support for DTX); do { *bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], (int16_t*) bitstream, my_rate_, num_20ms_frames); // increment the read index this tell the caller that how far // we have gone forward in reading the audio buffer in_audio_ix_read_ += 160; // sanity check if (*bitstream_len_byte < 0) { // error has happened WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "InternalEncode: Encode error for G729_1"); *bitstream_len_byte = 0; return -1; } num_encoded_samples += 160; } while (*bitstream_len_byte == 0); // This criteria will change if we have Annex C. if (*bitstream_len_byte != byte_length_frame) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "InternalEncode: Encode error for G729_1"); *bitstream_len_byte = 0; return -1; } if (num_encoded_samples != frame_len_smpl_) { *bitstream_len_byte = 0; return -1; } return *bitstream_len_byte; } int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */, int16_t /* bitstream_len_byte */, int16_t* /* audio */, int16_t* /* audio_samples */, int8_t* /* speech_type */) { return 0; } int16_t ACMG729_1::InternalInitEncoder( WebRtcACMCodecParams* codec_params) { //set the bit rate and initialize my_rate_ = codec_params->codec_inst.rate; return SetBitRateSafe((uint32_t) my_rate_); } int16_t ACMG729_1::InternalInitDecoder( WebRtcACMCodecParams* /* codec_params */) { if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "InternalInitDecoder: init decoder failed for G729_1"); return -1; } return 0; } int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def, const CodecInst& codec_inst) { if (!decoder_initialized_) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "CodeDef: Decoder uninitialized for G729_1"); return -1; } // Fill up the structure by calling // "SET_CODEC_PAR" & "SET_G729_FUNCTION." // Then call NetEQ to add the codec to it's // database. SET_CODEC_PAR((codec_def), kDecoderG729_1, codec_inst.pltype, decoder_inst_ptr_, 16000); SET_G729_1_FUNCTIONS((codec_def)); return 0; } ACMGenericCodec* ACMG729_1::CreateInstance(void) { return NULL; } int16_t ACMG729_1::InternalCreateEncoder() { if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "InternalCreateEncoder: create encoder failed for G729_1"); return -1; } return 0; } void ACMG729_1::DestructEncoderSafe() { encoder_exist_ = false; encoder_initialized_ = false; if (encoder_inst_ptr_ != NULL) { WebRtcG7291_Free(encoder_inst_ptr_); encoder_inst_ptr_ = NULL; } } int16_t ACMG729_1::InternalCreateDecoder() { if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "InternalCreateDecoder: create decoder failed for G729_1"); return -1; } return 0; } void ACMG729_1::DestructDecoderSafe() { decoder_exist_ = false; decoder_initialized_ = false; if (decoder_inst_ptr_ != NULL) { WebRtcG7291_Free(decoder_inst_ptr_); decoder_inst_ptr_ = NULL; } } void ACMG729_1::InternalDestructEncoderInst(void* ptr_inst) { if (ptr_inst != NULL) { // WebRtcG7291_Free((G729_1_inst_t*)ptrInst); } return; } int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) { // allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000, // 22000, 24000, 26000, 28000, 30000, 32000}; // TODO(tlegrand): This check exists in one other place two. Should be // possible to reuse code. switch (rate) { case 8000: { my_rate_ = 8000; break; } case 12000: { my_rate_ = 12000; break; } case 14000: { my_rate_ = 14000; break; } case 16000: { my_rate_ = 16000; break; } case 18000: { my_rate_ = 18000; break; } case 20000: { my_rate_ = 20000; break; } case 22000: { my_rate_ = 22000; break; } case 24000: { my_rate_ = 24000; break; } case 26000: { my_rate_ = 26000; break; } case 28000: { my_rate_ = 28000; break; } case 30000: { my_rate_ = 30000; break; } case 32000: { my_rate_ = 32000; break; } default: { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_, "SetBitRateSafe: Invalid rate G729_1"); return -1; } } // Re-init with new rate if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_, flag_g729_mode_) >= 0) { encoder_params_.codec_inst.rate = my_rate_; return 0; } else { return -1; } } #endif } // namespace webrtc