
This effectively reverts r8211. The problem with r8210 was that the change in constructor signature was not done for other codec selections that then default one. That is, some code that was hidden under #ifdef did not get updated. This is now fixed. BUG=4228 COAUTHOR=kwiberg@webrtc.org TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37879004 Cr-Commit-Position: refs/heads/master@{#8215} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8215 4adac7df-926f-26a2-2b94-8c16560cd09d
238 lines
6.2 KiB
C++
238 lines
6.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/main/acm2/acm_g7291.h"
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#ifdef WEBRTC_CODEC_G729_1
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// NOTE! G.729.1 is not included in the open-source package. Modify this file
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// or your codec API to match the function calls and names of used G.729.1 API
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// file.
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#include "webrtc/modules/audio_coding/main/codecs/g7291/interface/g7291_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#endif
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namespace webrtc {
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namespace acm2 {
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#ifndef WEBRTC_CODEC_G729_1
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ACMG729_1::ACMG729_1(int16_t /* codec_id */, bool enable_red)
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: ACMGenericCodec(enable_red),
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encoder_inst_ptr_(NULL),
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my_rate_(32000),
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flag_8khz_(0),
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flag_g729_mode_(0) {
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return;
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}
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ACMG729_1::~ACMG729_1() { return; }
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int16_t ACMG729_1::InternalEncode(uint8_t* /* bitstream */,
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int16_t* /* bitstream_len_byte */) {
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return -1;
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}
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int16_t ACMG729_1::InternalInitEncoder(
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WebRtcACMCodecParams* /* codec_params */) {
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return -1;
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}
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ACMGenericCodec* ACMG729_1::CreateInstance(void) { return NULL; }
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int16_t ACMG729_1::InternalCreateEncoder() { return -1; }
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void ACMG729_1::DestructEncoderSafe() { return; }
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int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) { return -1; }
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#else //===================== Actual Implementation =======================
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struct G729_1_inst_t_;
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ACMG729_1::ACMG729_1(int16_t codec_id, bool enable_red)
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: ACMGenericCodec(enable_red),
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encoder_inst_ptr_(NULL),
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my_rate_(32000), // Default rate.
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flag_8khz_(0),
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flag_g729_mode_(0) {
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// TODO(tlegrand): We should add codec_id as a input variable to the
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// constructor of ACMGenericCodec.
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codec_id_ = codec_id;
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return;
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}
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ACMG729_1::~ACMG729_1() {
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if (encoder_inst_ptr_ != NULL) {
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WebRtcG7291_Free(encoder_inst_ptr_);
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encoder_inst_ptr_ = NULL;
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}
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return;
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}
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int16_t ACMG729_1::InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) {
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// Initialize before entering the loop
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int16_t num_encoded_samples = 0;
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*bitstream_len_byte = 0;
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int16_t byte_length_frame = 0;
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// Derive number of 20ms frames per encoded packet.
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// [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
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int16_t num_20ms_frames = (frame_len_smpl_ / 320);
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// Byte length for the frame. +1 is for rate information.
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byte_length_frame =
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my_rate_ / (8 * 50) * num_20ms_frames + (1 - flag_g729_mode_);
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// The following might be revised if we have G729.1 Annex C (support for DTX);
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do {
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*bitstream_len_byte = WebRtcG7291_Encode(
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encoder_inst_ptr_, &in_audio_[in_audio_ix_read_],
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reinterpret_cast<int16_t*>(bitstream), my_rate_, num_20ms_frames);
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// increment the read index this tell the caller that how far
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// we have gone forward in reading the audio buffer
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in_audio_ix_read_ += 160;
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// sanity check
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if (*bitstream_len_byte < 0) {
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// error has happened
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"InternalEncode: Encode error for G729_1");
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*bitstream_len_byte = 0;
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return -1;
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}
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num_encoded_samples += 160;
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} while (*bitstream_len_byte == 0);
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// This criteria will change if we have Annex C.
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if (*bitstream_len_byte != byte_length_frame) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"InternalEncode: Encode error for G729_1");
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*bitstream_len_byte = 0;
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return -1;
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}
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if (num_encoded_samples != frame_len_smpl_) {
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*bitstream_len_byte = 0;
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return -1;
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}
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return *bitstream_len_byte;
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}
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int16_t ACMG729_1::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
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// set the bit rate and initialize
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my_rate_ = codec_params->codec_inst.rate;
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return SetBitRateSafe((uint32_t)my_rate_);
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}
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ACMGenericCodec* ACMG729_1::CreateInstance(void) { return NULL; }
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int16_t ACMG729_1::InternalCreateEncoder() {
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if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError,
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webrtc::kTraceAudioCoding,
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unique_id_,
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"InternalCreateEncoder: create encoder failed for G729_1");
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return -1;
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}
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return 0;
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}
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void ACMG729_1::DestructEncoderSafe() {
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encoder_exist_ = false;
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encoder_initialized_ = false;
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if (encoder_inst_ptr_ != NULL) {
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WebRtcG7291_Free(encoder_inst_ptr_);
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encoder_inst_ptr_ = NULL;
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}
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}
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int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
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// allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
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// 22000, 24000, 26000, 28000, 30000, 32000};
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// TODO(tlegrand): This check exists in one other place two. Should be
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// possible to reuse code.
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switch (rate) {
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case 8000: {
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my_rate_ = 8000;
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break;
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}
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case 12000: {
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my_rate_ = 12000;
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break;
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}
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case 14000: {
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my_rate_ = 14000;
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break;
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}
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case 16000: {
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my_rate_ = 16000;
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break;
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}
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case 18000: {
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my_rate_ = 18000;
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break;
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}
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case 20000: {
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my_rate_ = 20000;
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break;
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}
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case 22000: {
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my_rate_ = 22000;
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break;
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}
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case 24000: {
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my_rate_ = 24000;
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break;
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}
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case 26000: {
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my_rate_ = 26000;
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break;
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}
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case 28000: {
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my_rate_ = 28000;
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break;
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}
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case 30000: {
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my_rate_ = 30000;
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break;
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}
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case 32000: {
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my_rate_ = 32000;
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break;
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}
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default: {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"SetBitRateSafe: Invalid rate G729_1");
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return -1;
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}
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}
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// Re-init with new rate
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if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_,
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flag_g729_mode_) >= 0) {
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encoder_params_.codec_inst.rate = my_rate_;
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return 0;
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} else {
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return -1;
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}
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}
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#endif
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} // namespace acm2
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} // namespace webrtc
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