Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

@@ -0,0 +1,181 @@
+/*
+ *  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 "modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "opus.h"
+
+#include "common_audio/signal_processing/resample_by_2_internal.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+enum {
+  /* We always produce 20ms frames. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 20,
+
+  /* The format allows up to 120ms frames. Since we
+   * don't control the other side, we must allow
+   * for packets that large. NetEq is currently
+   * limited to 60 ms on the receive side.
+   */
+  kWebRtcOpusMaxDecodeFrameSizeMs = 120,
+
+  /* Sample count is 48 kHz * samples per frame. */
+  kWebRtcOpusMaxFrameSize = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
+};
+
+struct WebRtcOpusEncInst {
+  OpusEncoder* encoder;
+};
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
+  OpusEncInst* state;
+  state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
+  if (state) {
+    int error;
+    state->encoder = opus_encoder_create(48000, channels, OPUS_APPLICATION_VOIP,
+                                         &error);
+    if (error == OPUS_OK || state->encoder != NULL ) {
+      *inst = state;
+      return 0;
+    }
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
+  opus_encoder_destroy(inst->encoder);
+  return 0;
+}
+
+int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
+                          int16_t length_encoded_buffer, uint8_t* encoded) {
+  opus_int16* audio = (opus_int16*) audio_in;
+  unsigned char* coded = encoded;
+  int res;
+
+  if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
+    return -1;
+  }
+
+  res = opus_encode(inst->encoder, audio, samples, coded,
+                    length_encoded_buffer);
+
+  if (res > 0) {
+    return res;
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
+  return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
+}
+
+struct WebRtcOpusDecInst {
+  int16_t state_48_32[8];
+  OpusDecoder* decoder;
+};
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+  OpusDecInst* state;
+  state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
+  if (state) {
+    int error;
+    // Always create a 48000 Hz Opus decoder.
+    state->decoder = opus_decoder_create(48000, channels, &error);
+    if (error == OPUS_OK && state->decoder != NULL ) {
+      *inst = state;
+      return 0;
+    }
+    free(state);
+    state = NULL;
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
+  opus_decoder_destroy(inst->decoder);
+  free(inst);
+  return 0;
+}
+
+int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  if (error == OPUS_OK) {
+    memset(inst->state_48_32, 0, sizeof(inst->state_48_32));
+    return 0;
+  }
+  return -1;
+}
+
+static int DecodeNative(OpusDecInst* inst, int16_t* encoded,
+                        int16_t encoded_bytes, int16_t* decoded,
+                        int16_t* audio_type) {
+  unsigned char* coded = (unsigned char*) encoded;
+  opus_int16* audio = (opus_int16*) decoded;
+
+  int res = opus_decode(inst->decoder, coded, encoded_bytes, audio,
+                        kWebRtcOpusMaxFrameSize, 0);
+  /* TODO(tlegrand): set to DTX for zero-length packets? */
+  *audio_type = 0;
+
+  if (res > 0) {
+    return res;
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
+                          int16_t encoded_bytes, int16_t* decoded,
+                          int16_t* audio_type) {
+  /* Enough for 120 ms (the largest Opus packet size) of mono audio at 48 kHz
+   * and resampler overlap. This will need to be enlarged for stereo decoding.
+   */
+  int16_t buffer16[kWebRtcOpusMaxFrameSize];
+  int32_t buffer32[kWebRtcOpusMaxFrameSize + 7];
+  int decoded_samples;
+  int blocks;
+  int16_t output_samples;
+  int i;
+
+  /* Decode to a temporary buffer. */
+  decoded_samples = DecodeNative(inst, encoded, encoded_bytes, buffer16,
+                                 audio_type);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+  /* Resample from 48 kHz to 32 kHz. */
+  for (i = 0; i < 7; i++) {
+    buffer32[i] = inst->state_48_32[i];
+    inst->state_48_32[i] = buffer16[decoded_samples -7 + i];
+  }
+  for (i = 0; i < decoded_samples; i++) {
+    buffer32[7 + i] = buffer16[i];
+  }
+  /* Resampling 3 samples to 2. Function divides the input in |blocks| number
+   * of 3-sample groups, and output is |blocks| number of 2-sample groups. */
+  blocks = decoded_samples / 3;
+  WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks);
+  output_samples = (int16_t) (blocks * 2);
+  WebRtcSpl_VectorBitShiftW32ToW16(decoded, output_samples, buffer32, 15);
+
+  return output_samples;
+}
+
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                             int16_t number_of_lost_frames) {
+  /* TODO(tlegrand): We can pass NULL to opus_decode to activate packet
+   * loss concealment, but I don't know how many samples
+   * number_of_lost_frames corresponds to. */
+  return -1;
+}