Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 1569caf..f8142e2 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
#include <algorithm>
-#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"
@@ -28,24 +27,27 @@
// 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,
- uint32_t* encoded_timestamp) {
+ // |encoded|, and provides the number of encoded bytes in |encoded_bytes|.
+ // In case of error, false is returned, otherwise true. It is an error for the
+ // encoder to attempt to produce more than |max_encoded_bytes| bytes of
+ // output.
+ bool Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t num_samples,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) {
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, encoded_timestamp);
- 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;
+ bool ret = Encode(timestamp,
+ audio,
+ max_encoded_bytes,
+ encoded,
+ encoded_bytes,
+ encoded_timestamp);
+ CHECK_LE(*encoded_bytes, max_encoded_bytes);
+ return ret;
}
// Returns the input sample rate in Hz, the number of input channels, and the
@@ -56,11 +58,12 @@
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,
- uint32_t* encoded_timestamp) = 0;
+ virtual bool Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) = 0;
};
} // namespace webrtc