From cb858ba3974c921627e81805e2ab4a2ae52c6619 Mon Sep 17 00:00:00 2001 From: "kwiberg@webrtc.org" Date: Mon, 8 Dec 2014 17:11:44 +0000 Subject: [PATCH] Make an AudioEncoder subclass for iLBC BUG=3926 R=henrik.lundin@webrtc.org, kjellander@google.com TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32649005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/audio_coding/BUILD.gn | 2 + .../codecs/ilbc/audio_encoder_ilbc.cc | 95 +++++++++++++++++++ .../modules/audio_coding/codecs/ilbc/ilbc.c | 9 +- .../audio_coding/codecs/ilbc/ilbc.gypi | 2 + .../ilbc/interface/audio_encoder_ilbc.h | 55 +++++++++++ .../audio_coding/codecs/ilbc/interface/ilbc.h | 2 +- .../audio_coding/codecs/ilbc/test/iLBC_test.c | 3 +- .../audio_coding/main/acm2/acm_ilbc.cc | 2 +- .../neteq/audio_decoder_unittest.cc | 27 +----- .../audio_coding/neteq/test/RTPencode.cc | 3 +- 10 files changed, 171 insertions(+), 29 deletions(-) create mode 100644 webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc create mode 100644 webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 0c087ecef..184d37e28 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -198,6 +198,8 @@ config("ilbc_config") { source_set("ilbc") { sources = [ + "codecs/ilbc/audio_encoder_ilbc.cc", + "codecs/ilbc/include/audio_encoder_ilbc.h", "codecs/ilbc/abs_quant.c", "codecs/ilbc/abs_quant.h", "codecs/ilbc/abs_quant_loop.c", diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc new file mode 100644 index 000000000..cf1d63226 --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc @@ -0,0 +1,95 @@ +/* + * Copyright (c) 2014 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/codecs/ilbc/interface/audio_encoder_ilbc.h" + +#include +#include +#include "webrtc/base/checks.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" + +namespace webrtc { + +namespace { + +const int kSampleRateHz = 8000; + +} // namespace + +AudioEncoderIlbc::AudioEncoderIlbc(const Config& config) + : payload_type_(config.payload_type), + num_10ms_frames_per_packet_(config.frame_size_ms / 10), + num_10ms_frames_buffered_(0) { + CHECK(config.frame_size_ms == 20 || config.frame_size_ms == 30) + << "Frame size must be 20 or 30 ms."; + DCHECK_LE(kSampleRateHz / 100 * num_10ms_frames_per_packet_, + kMaxSamplesPerPacket); + CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_)); + CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, config.frame_size_ms)); +} + +AudioEncoderIlbc::~AudioEncoderIlbc() { + CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_)); +} + +int AudioEncoderIlbc::sample_rate_hz() const { + return kSampleRateHz; +} +int AudioEncoderIlbc::num_channels() const { + return 1; +} +int AudioEncoderIlbc::Num10MsFramesInNextPacket() const { + return num_10ms_frames_per_packet_; +} + +bool AudioEncoderIlbc::EncodeInternal(uint32_t timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded, + size_t* encoded_bytes, + EncodedInfo* info) { + const size_t expected_output_len ATTRIBUTE_UNUSED = + num_10ms_frames_per_packet_ == 2 ? 38 : 50; + DCHECK_GE(max_encoded_bytes, expected_output_len); + + // Save timestamp if starting a new packet. + if (num_10ms_frames_buffered_ == 0) + first_timestamp_in_buffer_ = timestamp; + + // Buffer input. + std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_, + audio, + kSampleRateHz / 100 * sizeof(audio[0])); + + // If we don't yet have enough buffered input for a whole packet, we're done + // for now. + if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { + *encoded_bytes = 0; + return true; + } + + // Encode buffered input. + DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); + num_10ms_frames_buffered_ = 0; + const int output_len = WebRtcIlbcfix_Encode( + encoder_, + input_buffer_, + kSampleRateHz / 100 * num_10ms_frames_per_packet_, + encoded); + if (output_len == -1) + return false; // Encoding error. + DCHECK_EQ(output_len, static_cast(expected_output_len)); + *encoded_bytes = output_len; + info->encoded_timestamp = first_timestamp_in_buffer_; + info->payload_type = payload_type_; + return true; +} + +} // namespace webrtc diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c index 21d159f58..97b317460 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c +++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c @@ -86,8 +86,10 @@ int16_t WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst, int16_t mode) } } -int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, const int16_t *speechIn, int16_t len, int16_t *encoded) { - +int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t* iLBCenc_inst, + const int16_t* speechIn, + int16_t len, + uint8_t* encoded) { int16_t pos = 0; int16_t encpos = 0; @@ -104,7 +106,8 @@ int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, const int16_t *speech /* call encoder */ while (possection == 0) diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi index ec3284f82..dcee4be90 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi +++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi @@ -25,9 +25,11 @@ ], }, 'sources': [ + 'interface/audio_encoder_ilbc.h', 'interface/ilbc.h', 'abs_quant.c', 'abs_quant_loop.c', + 'audio_encoder_ilbc.cc', 'augmented_cb_corr.c', 'bw_expand.c', 'cb_construct.c', diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h new file mode 100644 index 000000000..199ef2b1f --- /dev/null +++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h @@ -0,0 +1,55 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_ + +#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +class AudioEncoderIlbc : public AudioEncoder { + public: + struct Config { + Config() : payload_type(102), frame_size_ms(30) {} + + int payload_type; + int frame_size_ms; + }; + + explicit AudioEncoderIlbc(const Config& config); + virtual ~AudioEncoderIlbc(); + + virtual int sample_rate_hz() const OVERRIDE; + virtual int num_channels() const OVERRIDE; + virtual int Num10MsFramesInNextPacket() const OVERRIDE; + + protected: + virtual bool EncodeInternal(uint32_t timestamp, + const int16_t* audio, + size_t max_encoded_bytes, + uint8_t* encoded, + size_t* encoded_bytes, + EncodedInfo* info) OVERRIDE; + + private: + static const int kMaxSamplesPerPacket = 240; + const int payload_type_; + const int num_10ms_frames_per_packet_; + int num_10ms_frames_buffered_; + uint32_t first_timestamp_in_buffer_; + int16_t input_buffer_[kMaxSamplesPerPacket]; + iLBC_encinst_t* encoder_; +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_ diff --git a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h index ccb3f5e3e..8e56008fc 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h +++ b/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h @@ -138,7 +138,7 @@ extern "C" { int16_t WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, const int16_t *speechIn, int16_t len, - int16_t *encoded); + uint8_t* encoded); /**************************************************************************** * WebRtcIlbcfix_DecoderInit(...) diff --git a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c index 4b86b9106..4c1b5cbaa 100644 --- a/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c +++ b/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c @@ -163,7 +163,8 @@ int main(int argc, char* argv[]) /* encoding */ fprintf(stderr, "--- Encoding block %i --- ",blockcount); - len=WebRtcIlbcfix_Encode(Enc_Inst, data, (int16_t)frameLen, encoded_data); + len=WebRtcIlbcfix_Encode(Enc_Inst, data, (int16_t)frameLen, + (uint8_t*)encoded_data); fprintf(stderr, "\r"); /* write byte file */ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc index d23ec6edc..94d655f18 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc @@ -61,7 +61,7 @@ int16_t ACMILBC::InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte) { *bitstream_len_byte = WebRtcIlbcfix_Encode( encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], frame_len_smpl_, - reinterpret_cast(bitstream)); + bitstream); if (*bitstream_len_byte < 0) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc index bbcf9ed3e..2e8118df4 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -23,7 +23,7 @@ #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" #include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h" #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h" -#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h" #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h" #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h" @@ -324,25 +324,10 @@ class AudioDecoderIlbcTest : public AudioDecoderTest { data_length_ = 10 * frame_size_; decoder_ = new AudioDecoderIlbc; assert(decoder_); - WebRtcIlbcfix_EncoderCreate(&encoder_); - } - - ~AudioDecoderIlbcTest() { - WebRtcIlbcfix_EncoderFree(encoder_); - } - - virtual void InitEncoder() { - ASSERT_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, 30)); // 30 ms. - } - - virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, - uint8_t* output) { - int enc_len_bytes = - WebRtcIlbcfix_Encode(encoder_, input, - static_cast(input_len_samples), - reinterpret_cast(output)); - EXPECT_EQ(50, enc_len_bytes); - return enc_len_bytes; + AudioEncoderIlbc::Config config; + config.frame_size_ms = 30; + config.payload_type = payload_type_; + audio_encoder_.reset(new AudioEncoderIlbc(config)); } // Overload the default test since iLBC's function WebRtcIlbcfix_NetEqPlc does @@ -362,8 +347,6 @@ class AudioDecoderIlbcTest : public AudioDecoderTest { // Simply call DecodePlc and verify that we get 0 as return value. EXPECT_EQ(0, decoder_->DecodePlc(1, output.get())); } - - iLBC_encinst_t* encoder_; }; class AudioDecoderIsacFloatTest : public AudioDecoderTest { diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc index efd006915..49d66e932 100644 --- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc +++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc @@ -1621,7 +1621,8 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e #endif #ifdef CODEC_ILBC else if (coder==webrtc::kDecoderILBC) { /*iLBC */ - cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(int16_t*)encoded); + cdlen = WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, + frameLen, encoded); } #endif #if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC