| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "remoting/codec/audio_encoder_speex.h" |
| |
| #include <string> |
| #include <sstream> |
| |
| #include "base/basictypes.h" |
| #include "base/logging.h" |
| #include "base/stl_util.h" |
| #include "remoting/proto/audio.pb.h" |
| #include "third_party/speex/include/speex/speex_callbacks.h" |
| #include "third_party/speex/include/speex/speex_stereo.h" |
| |
| namespace { |
| // A quality of 8 in wide band mode corresponds to 27,800 bits per second. |
| const int kSpeexHighQuality = 8; |
| const int kEncodedDataBufferSize = 0xFF; |
| } // namespace |
| |
| namespace remoting { |
| |
| AudioEncoderSpeex::AudioEncoderSpeex() |
| : leftover_frames_(0) { |
| // Create and initialize the Speex structures. |
| speex_bits_.reset(new SpeexBits()); |
| speex_bits_init(speex_bits_.get()); |
| speex_state_ = speex_encoder_init(&speex_wb_mode); |
| |
| // Set the encoding quality. |
| int quality = kSpeexHighQuality; |
| speex_encoder_ctl(speex_state_, SPEEX_SET_QUALITY, &quality); |
| |
| // Get the frame size and construct the input buffer accordingly. |
| int result = speex_encoder_ctl(speex_state_, |
| SPEEX_GET_FRAME_SIZE, |
| &speex_frame_size_); |
| CHECK_EQ(result, 0); |
| |
| leftover_buffer_.reset( |
| new int16[speex_frame_size_ * AudioPacket::CHANNELS_STEREO]); |
| } |
| |
| AudioEncoderSpeex::~AudioEncoderSpeex() { |
| speex_encoder_destroy(speex_state_); |
| speex_bits_destroy(speex_bits_.get()); |
| } |
| |
| scoped_ptr<AudioPacket> AudioEncoderSpeex::Encode( |
| scoped_ptr<AudioPacket> packet) { |
| DCHECK_EQ(AudioPacket::ENCODING_RAW, packet->encoding()); |
| DCHECK_EQ(1, packet->data_size()); |
| DCHECK_EQ(AudioPacket::BYTES_PER_SAMPLE_2, packet->bytes_per_sample()); |
| DCHECK_NE(AudioPacket::SAMPLING_RATE_INVALID, packet->sampling_rate()); |
| DCHECK_EQ(AudioPacket::CHANNELS_STEREO, packet->channels()); |
| |
| int frames_left = |
| packet->data(0).size() / packet->bytes_per_sample() / packet->channels(); |
| const int16* next_sample = |
| reinterpret_cast<const int16*>(packet->data(0).data()); |
| |
| // Create a new packet of encoded data. |
| scoped_ptr<AudioPacket> encoded_packet(new AudioPacket()); |
| encoded_packet->set_encoding(AudioPacket::ENCODING_SPEEX); |
| encoded_packet->set_sampling_rate(packet->sampling_rate()); |
| encoded_packet->set_bytes_per_sample(packet->bytes_per_sample()); |
| encoded_packet->set_channels(packet->channels()); |
| |
| while (leftover_frames_ + frames_left >= speex_frame_size_) { |
| int16* unencoded_buffer = NULL; |
| int frames_consumed = 0; |
| |
| if (leftover_frames_ > 0) { |
| unencoded_buffer = leftover_buffer_.get(); |
| frames_consumed = speex_frame_size_ - leftover_frames_; |
| |
| memcpy(leftover_buffer_.get() + leftover_frames_ * packet->channels(), |
| next_sample, |
| frames_consumed * packet->bytes_per_sample() * packet->channels()); |
| |
| leftover_frames_ = 0; |
| } else { |
| unencoded_buffer = const_cast<int16*>(next_sample); |
| frames_consumed = speex_frame_size_; |
| } |
| |
| // Transform stereo to mono. |
| speex_encode_stereo_int(unencoded_buffer, |
| speex_frame_size_, |
| speex_bits_.get()); |
| |
| // Encode the frame, treating all samples as integers. |
| speex_encode_int(speex_state_, |
| unencoded_buffer, |
| speex_bits_.get()); |
| |
| next_sample += frames_consumed * packet->channels(); |
| frames_left -= frames_consumed; |
| |
| std::string* new_data = encoded_packet->add_data(); |
| new_data->resize(speex_bits_nbytes(speex_bits_.get())); |
| |
| // Copy the encoded data from the bits structure into the buffer. |
| int bytes_written = speex_bits_write(speex_bits_.get(), |
| string_as_array(new_data), |
| new_data->size()); |
| |
| // Expect that the bytes are all written. |
| DCHECK_EQ(bytes_written, static_cast<int>(new_data->size())); |
| |
| // Reset the bits structure for this frame. |
| speex_bits_reset(speex_bits_.get()); |
| } |
| |
| // Store the leftover samples. |
| if (frames_left > 0) { |
| CHECK_LE(leftover_frames_ + frames_left, speex_frame_size_); |
| memcpy(leftover_buffer_.get() + leftover_frames_ * packet->channels(), |
| next_sample, |
| frames_left * packet->bytes_per_sample() * packet->channels()); |
| leftover_frames_ += frames_left; |
| } |
| |
| // Return NULL if there's nothing in the packet. |
| if (encoded_packet->data_size() == 0) |
| return scoped_ptr<AudioPacket>(); |
| |
| return encoded_packet.Pass(); |
| } |
| |
| } // namespace remoting |