diff options
Diffstat (limited to 'webrtc/modules/audio_coding/neteq/neteq_impl.h')
| -rw-r--r-- | webrtc/modules/audio_coding/neteq/neteq_impl.h | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h index c001e53b81..940deadd2f 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_impl.h +++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h @@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ +#include <string> + #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" @@ -79,8 +81,7 @@ class NetEqImpl : public webrtc::NetEq { // the same tick rate as the RTP timestamp of the current payload. // Returns 0 on success, -1 on failure. int InsertPacket(const WebRtcRTPHeader& rtp_header, - const uint8_t* payload, - size_t length_bytes, + rtc::ArrayView<const uint8_t> payload, uint32_t receive_timestamp) override; // Inserts a sync-packet into packet queue. Sync-packets are decoded to @@ -106,20 +107,16 @@ class NetEqImpl : public webrtc::NetEq { int GetAudio(size_t max_length, int16_t* output_audio, size_t* samples_per_channel, - int* num_channels, + size_t* num_channels, NetEqOutputType* type) override; - // Associates |rtp_payload_type| with |codec| and stores the information in - // the codec database. Returns kOK on success, kFail on failure. int RegisterPayloadType(NetEqDecoder codec, + const std::string& codec_name, uint8_t rtp_payload_type) override; - // Provides an externally created decoder object |decoder| to insert in the - // decoder database. The decoder implements a decoder of type |codec| and - // associates it with |rtp_payload_type|. The decoder will produce samples - // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure. int RegisterExternalDecoder(AudioDecoder* decoder, NetEqDecoder codec, + const std::string& codec_name, uint8_t rtp_payload_type, int sample_rate_hz) override; @@ -169,6 +166,8 @@ class NetEqImpl : public webrtc::NetEq { bool GetPlayoutTimestamp(uint32_t* timestamp) override; + int last_output_sample_rate_hz() const override; + int SetTargetNumberOfChannels() override; int SetTargetSampleRate() override; @@ -207,8 +206,7 @@ class NetEqImpl : public webrtc::NetEq { // above. Returns 0 on success, otherwise an error code. // TODO(hlundin): Merge this with InsertPacket above? int InsertPacketInternal(const WebRtcRTPHeader& rtp_header, - const uint8_t* payload, - size_t length_bytes, + rtc::ArrayView<const uint8_t> payload, uint32_t receive_timestamp, bool is_sync_packet) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); @@ -222,7 +220,8 @@ class NetEqImpl : public webrtc::NetEq { int GetAudioInternal(size_t max_length, int16_t* output, size_t* samples_per_channel, - int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); + size_t* num_channels) + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); // Provides a decision to the GetAudioInternal method. The decision what to // do is written to |operation|. Packets to decode are written to @@ -377,6 +376,7 @@ class NetEqImpl : public webrtc::NetEq { StatisticsCalculator stats_ GUARDED_BY(crit_sect_); int fs_hz_ GUARDED_BY(crit_sect_); int fs_mult_ GUARDED_BY(crit_sect_); + int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_); size_t output_size_samples_ GUARDED_BY(crit_sect_); size_t decoder_frame_length_ GUARDED_BY(crit_sect_); Modes last_mode_ GUARDED_BY(crit_sect_); |