New interface class AudioEncoder

This class will be the base for new C++ wrapper classes for all
encoders.

BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7463 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 64 insertions, 0 deletions

diff --git a/modules/audio_coding/codecs/audio_encoder.h b/modules/audio_coding/codecs/audio_encoder.h
new file mode 100644
index 00000000..0d446bcb
--- /dev/null
+++ b/modules/audio_coding/codecs/audio_encoder.h
@@ -0,0 +1,64 @@
+/*
+ *  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_AUDIO_ENCODER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// This is the interface class for encoders in AudioCoding module. Each codec
+// codec type must have an implementation of this class.
+class AudioEncoder {
+ public:
+  virtual ~AudioEncoder() {}
+
+  // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
+  // num_channels() samples). Multi-channel audio must be sample-interleaved.
+  // If successful, the encoder produces zero or more bytes of output in
+  // |encoded|, and returns the number of bytes. In case of error, -1 is
+  // returned. It is an error for the encoder to attempt to produce more than
+  // |max_encoded_bytes| bytes of output.
+  ssize_t Encode(uint32_t timestamp,
+                 const int16_t* audio,
+                 size_t num_samples,
+                 size_t max_encoded_bytes,
+                 uint8_t* encoded) {
+    CHECK_EQ(num_samples,
+             static_cast<size_t>(sample_rate_hz() / 100 * num_channels()));
+    ssize_t num_bytes = Encode(timestamp, audio, max_encoded_bytes, encoded);
+    CHECK_LE(num_bytes,
+             static_cast<ssize_t>(std::min(
+                 max_encoded_bytes,
+                 static_cast<size_t>(std::numeric_limits<ssize_t>::max()))));
+    return num_bytes;
+  }
+
+  // Returns the input sample rate in Hz, the number of input channels, and the
+  // number of 10 ms frames the encoder puts in one output packet. These are
+  // constants set at instantiation time.
+  virtual int sample_rate_hz() const = 0;
+  virtual int num_channels() const = 0;
+  virtual int num_10ms_frames_per_packet() const = 0;
+
+ protected:
+  virtual ssize_t Encode(uint32_t timestamp,
+                         const int16_t* audio,
+                         size_t max_encoded_bytes,
+                         uint8_t* encoded) = 0;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_