diff options
Diffstat (limited to 'webrtc/modules/audio_coding/codecs/audio_encoder.h')
| -rw-r--r-- | webrtc/modules/audio_coding/codecs/audio_encoder.h | 10 |
1 files changed, 5 insertions, 5 deletions
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h index cda9d86f2e..a46b0e86a7 100644 --- a/webrtc/modules/audio_coding/codecs/audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h @@ -14,6 +14,7 @@ #include <algorithm> #include <vector> +#include "webrtc/base/array_view.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -60,7 +61,7 @@ class AudioEncoder { // Returns the input sample rate in Hz and the number of input channels. // These are constants set at instantiation time. virtual int SampleRateHz() const = 0; - virtual int NumChannels() const = 0; + virtual size_t NumChannels() const = 0; // Returns the rate at which the RTP timestamps are updated. The default // implementation returns SampleRateHz(). @@ -91,13 +92,12 @@ class AudioEncoder { // Encode() checks some preconditions, calls EncodeInternal() which does the // actual work, and then checks some postconditions. EncodedInfo Encode(uint32_t rtp_timestamp, - const int16_t* audio, - size_t num_samples_per_channel, + rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded); virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, + rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) = 0; @@ -125,7 +125,7 @@ class AudioEncoder { // Tells the encoder about the highest sample rate the decoder is expected to // use when decoding the bitstream. The encoder would typically use this // information to adjust the quality of the encoding. The default - // implementation just returns true. + // implementation does nothing. virtual void SetMaxPlaybackRate(int frequency_hz); // Tells the encoder what the projected packet loss rate is. The rate is in |