| Index: talk/media/base/mediachannel.h
|
| diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
|
| index ec11694e711f96b21658c663931da6be72bf5930..6c596831263f216b3dc6ae68b852d6e22affe232 100644
|
| --- a/talk/media/base/mediachannel.h
|
| +++ b/talk/media/base/mediachannel.h
|
| @@ -288,14 +288,14 @@ struct AudioOptions {
|
| Settable<bool> experimental_ns;
|
| Settable<bool> aec_dump;
|
| // Note that tx_agc_* only applies to non-experimental AGC.
|
| - Settable<uint16> tx_agc_target_dbov;
|
| - Settable<uint16> tx_agc_digital_compression_gain;
|
| + Settable<uint16_t> tx_agc_target_dbov;
|
| + Settable<uint16_t> tx_agc_digital_compression_gain;
|
| Settable<bool> tx_agc_limiter;
|
| - Settable<uint16> rx_agc_target_dbov;
|
| - Settable<uint16> rx_agc_digital_compression_gain;
|
| + Settable<uint16_t> rx_agc_target_dbov;
|
| + Settable<uint16_t> rx_agc_digital_compression_gain;
|
| Settable<bool> rx_agc_limiter;
|
| - Settable<uint32> recording_sample_rate;
|
| - Settable<uint32> playout_sample_rate;
|
| + Settable<uint32_t> recording_sample_rate;
|
| + Settable<uint32_t> playout_sample_rate;
|
| // Set DSCP value for packet sent from audio channel.
|
| Settable<bool> dscp;
|
| // Enable combined audio+bandwidth BWE.
|
| @@ -557,14 +557,14 @@ class MediaChannel : public sigslot::has_slots<> {
|
| // Removes an outgoing media stream.
|
| // ssrc must be the first SSRC of the media stream if the stream uses
|
| // multiple SSRCs.
|
| - virtual bool RemoveSendStream(uint32 ssrc) = 0;
|
| + virtual bool RemoveSendStream(uint32_t ssrc) = 0;
|
| // Creates a new incoming media stream with SSRCs and CNAME as described
|
| // by sp.
|
| virtual bool AddRecvStream(const StreamParams& sp) = 0;
|
| // Removes an incoming media stream.
|
| // ssrc must be the first SSRC of the media stream if the stream uses
|
| // multiple SSRCs.
|
| - virtual bool RemoveRecvStream(uint32 ssrc) = 0;
|
| + virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
|
|
|
| // Returns the absoulte sendtime extension id value from media channel.
|
| virtual int GetRtpSendTimeExtnId() const {
|
| @@ -640,7 +640,7 @@ struct SsrcSenderInfo {
|
| : ssrc(0),
|
| timestamp(0) {
|
| }
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| double timestamp; // NTP timestamp, represented as seconds since epoch.
|
| };
|
|
|
| @@ -649,7 +649,7 @@ struct SsrcReceiverInfo {
|
| : ssrc(0),
|
| timestamp(0) {
|
| }
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| double timestamp;
|
| };
|
|
|
| @@ -666,14 +666,14 @@ struct MediaSenderInfo {
|
| }
|
| // Temporary utility function for call sites that only provide SSRC.
|
| // As more info is added into SsrcSenderInfo, this function should go away.
|
| - void add_ssrc(uint32 ssrc) {
|
| + void add_ssrc(uint32_t ssrc) {
|
| SsrcSenderInfo stat;
|
| stat.ssrc = ssrc;
|
| add_ssrc(stat);
|
| }
|
| // Utility accessor for clients that are only interested in ssrc numbers.
|
| - std::vector<uint32> ssrcs() const {
|
| - std::vector<uint32> retval;
|
| + std::vector<uint32_t> ssrcs() const {
|
| + std::vector<uint32_t> retval;
|
| for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin();
|
| it != local_stats.end(); ++it) {
|
| retval.push_back(it->ssrc);
|
| @@ -683,14 +683,14 @@ struct MediaSenderInfo {
|
| // Utility accessor for clients that make the assumption only one ssrc
|
| // exists per media.
|
| // This will eventually go away.
|
| - uint32 ssrc() const {
|
| + uint32_t ssrc() const {
|
| if (local_stats.size() > 0) {
|
| return local_stats[0].ssrc;
|
| } else {
|
| return 0;
|
| }
|
| }
|
| - int64 bytes_sent;
|
| + int64_t bytes_sent;
|
| int packets_sent;
|
| int packets_lost;
|
| float fraction_lost;
|
| @@ -726,13 +726,13 @@ struct MediaReceiverInfo {
|
| }
|
| // Temporary utility function for call sites that only provide SSRC.
|
| // As more info is added into SsrcSenderInfo, this function should go away.
|
| - void add_ssrc(uint32 ssrc) {
|
| + void add_ssrc(uint32_t ssrc) {
|
| SsrcReceiverInfo stat;
|
| stat.ssrc = ssrc;
|
| add_ssrc(stat);
|
| }
|
| - std::vector<uint32> ssrcs() const {
|
| - std::vector<uint32> retval;
|
| + std::vector<uint32_t> ssrcs() const {
|
| + std::vector<uint32_t> retval;
|
| for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin();
|
| it != local_stats.end(); ++it) {
|
| retval.push_back(it->ssrc);
|
| @@ -742,7 +742,7 @@ struct MediaReceiverInfo {
|
| // Utility accessor for clients that make the assumption only one ssrc
|
| // exists per media.
|
| // This will eventually go away.
|
| - uint32 ssrc() const {
|
| + uint32_t ssrc() const {
|
| if (local_stats.size() > 0) {
|
| return local_stats[0].ssrc;
|
| } else {
|
| @@ -750,7 +750,7 @@ struct MediaReceiverInfo {
|
| }
|
| }
|
|
|
| - int64 bytes_rcvd;
|
| + int64_t bytes_rcvd;
|
| int packets_rcvd;
|
| int packets_lost;
|
| float fraction_lost;
|
| @@ -827,7 +827,7 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
|
| int decoding_cng;
|
| int decoding_plc_cng;
|
| // Estimated capture start time in NTP time in ms.
|
| - int64 capture_start_ntp_time_ms;
|
| + int64_t capture_start_ntp_time_ms;
|
| };
|
|
|
| struct VideoSenderInfo : public MediaSenderInfo {
|
| @@ -931,7 +931,7 @@ struct VideoReceiverInfo : public MediaReceiverInfo {
|
| int current_delay_ms;
|
|
|
| // Estimated capture start time in NTP time in ms.
|
| - int64 capture_start_ntp_time_ms;
|
| + int64_t capture_start_ntp_time_ms;
|
| };
|
|
|
| struct DataSenderInfo : public MediaSenderInfo {
|
| @@ -939,7 +939,7 @@ struct DataSenderInfo : public MediaSenderInfo {
|
| : ssrc(0) {
|
| }
|
|
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| };
|
|
|
| struct DataReceiverInfo : public MediaReceiverInfo {
|
| @@ -947,7 +947,7 @@ struct DataReceiverInfo : public MediaReceiverInfo {
|
| : ssrc(0) {
|
| }
|
|
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| };
|
|
|
| struct BandwidthEstimationInfo {
|
| @@ -1070,10 +1070,12 @@ class VoiceMediaChannel : public MediaChannel {
|
| // Starts or stops sending (and potentially capture) of local audio.
|
| virtual bool SetSend(SendFlags flag) = 0;
|
| // Configure stream for sending.
|
| - virtual bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
|
| + virtual bool SetAudioSend(uint32_t ssrc,
|
| + bool mute,
|
| + const AudioOptions* options,
|
| AudioRenderer* renderer) = 0;
|
| // Sets the renderer object to be used for the specified remote audio stream.
|
| - virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
|
| + virtual bool SetRemoteRenderer(uint32_t ssrc, AudioRenderer* renderer) = 0;
|
| // Gets current energy levels for all incoming streams.
|
| virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
|
| // Get the current energy level of the stream sent to the speaker.
|
| @@ -1085,11 +1087,11 @@ class VoiceMediaChannel : public MediaChannel {
|
| int cost_per_typing, int reporting_threshold, int penalty_decay,
|
| int type_event_delay) = 0;
|
| // Set left and right scale for speaker output volume of the specified ssrc.
|
| - virtual bool SetOutputScaling(uint32 ssrc, double left, double right) = 0;
|
| + virtual bool SetOutputScaling(uint32_t ssrc, double left, double right) = 0;
|
| // Specifies a ringback tone to be played during call setup.
|
| virtual bool SetRingbackTone(const char *buf, int len) = 0;
|
| // Plays or stops the aforementioned ringback tone
|
| - virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) = 0;
|
| + virtual bool PlayRingbackTone(uint32_t ssrc, bool play, bool loop) = 0;
|
| // Returns if the telephone-event has been negotiated.
|
| virtual bool CanInsertDtmf() { return false; }
|
| // Send and/or play a DTMF |event| according to the |flags|.
|
| @@ -1097,19 +1099,22 @@ class VoiceMediaChannel : public MediaChannel {
|
| // The |ssrc| should be either 0 or a valid send stream ssrc.
|
| // The valid value for the |event| are 0 to 15 which corresponding to
|
| // DTMF event 0-9, *, #, A-D.
|
| - virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) = 0;
|
| + virtual bool InsertDtmf(uint32_t ssrc,
|
| + int event,
|
| + int duration,
|
| + int flags) = 0;
|
| // Gets quality stats for the channel.
|
| virtual bool GetStats(VoiceMediaInfo* info) = 0;
|
| // Gets last reported error for this media channel.
|
| - virtual void GetLastMediaError(uint32* ssrc,
|
| + virtual void GetLastMediaError(uint32_t* ssrc,
|
| VoiceMediaChannel::Error* error) {
|
| ASSERT(error != NULL);
|
| *error = ERROR_NONE;
|
| }
|
|
|
| // Signal errors from MediaChannel. Arguments are:
|
| - // ssrc(uint32), and error(VoiceMediaChannel::Error).
|
| - sigslot::signal2<uint32, VoiceMediaChannel::Error> SignalMediaError;
|
| + // ssrc(uint32_t), and error(VoiceMediaChannel::Error).
|
| + sigslot::signal2<uint32_t, VoiceMediaChannel::Error> SignalMediaError;
|
| };
|
|
|
| struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
|
| @@ -1143,20 +1148,22 @@ class VideoMediaChannel : public MediaChannel {
|
| // Gets the currently set codecs/payload types to be used for outgoing media.
|
| virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
|
| // Sets the format of a specified outgoing stream.
|
| - virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) = 0;
|
| + virtual bool SetSendStreamFormat(uint32_t ssrc,
|
| + const VideoFormat& format) = 0;
|
| // Starts or stops playout of received video.
|
| virtual bool SetRender(bool render) = 0;
|
| // Starts or stops transmission (and potentially capture) of local video.
|
| virtual bool SetSend(bool send) = 0;
|
| // Configure stream for sending.
|
| - virtual bool SetVideoSend(uint32 ssrc, bool mute,
|
| + virtual bool SetVideoSend(uint32_t ssrc,
|
| + bool mute,
|
| const VideoOptions* options) = 0;
|
| // Sets the renderer object to be used for the specified stream.
|
| // If SSRC is 0, the renderer is used for the 'default' stream.
|
| - virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) = 0;
|
| + virtual bool SetRenderer(uint32_t ssrc, VideoRenderer* renderer) = 0;
|
| // If |ssrc| is 0, replace the default capturer (engine capturer) with
|
| // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC.
|
| - virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) = 0;
|
| + virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0;
|
| // Gets quality stats for the channel.
|
| virtual bool GetStats(VideoMediaInfo* info) = 0;
|
| // Send an intra frame to the receivers.
|
| @@ -1166,8 +1173,8 @@ class VideoMediaChannel : public MediaChannel {
|
| virtual void UpdateAspectRatio(int ratio_w, int ratio_h) = 0;
|
|
|
| // Signal errors from MediaChannel. Arguments are:
|
| - // ssrc(uint32), and error(VideoMediaChannel::Error).
|
| - sigslot::signal2<uint32, Error> SignalMediaError;
|
| + // ssrc(uint32_t), and error(VideoMediaChannel::Error).
|
| + sigslot::signal2<uint32_t, Error> SignalMediaError;
|
|
|
| protected:
|
| VideoRenderer *renderer_;
|
| @@ -1188,7 +1195,7 @@ enum DataMessageType {
|
| struct ReceiveDataParams {
|
| // The in-packet stream indentifier.
|
| // For SCTP, this is really SID, not SSRC.
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| // The type of message (binary, text, or control).
|
| DataMessageType type;
|
| // A per-stream value incremented per packet in the stream.
|
| @@ -1207,7 +1214,7 @@ struct ReceiveDataParams {
|
| struct SendDataParams {
|
| // The in-packet stream indentifier.
|
| // For SCTP, this is really SID, not SSRC.
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| // The type of message (binary, text, or control).
|
| DataMessageType type;
|
|
|
| @@ -1292,13 +1299,13 @@ class DataMediaChannel : public MediaChannel {
|
| const char*,
|
| size_t> SignalDataReceived;
|
| // Signal errors from MediaChannel. Arguments are:
|
| - // ssrc(uint32), and error(DataMediaChannel::Error).
|
| - sigslot::signal2<uint32, DataMediaChannel::Error> SignalMediaError;
|
| + // ssrc(uint32_t), and error(DataMediaChannel::Error).
|
| + sigslot::signal2<uint32_t, DataMediaChannel::Error> SignalMediaError;
|
| // Signal when the media channel is ready to send the stream. Arguments are:
|
| // writable(bool)
|
| sigslot::signal1<bool> SignalReadyToSend;
|
| // Signal for notifying that the remote side has closed the DataChannel.
|
| - sigslot::signal1<uint32> SignalStreamClosedRemotely;
|
| + sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
|
| };
|
|
|
| } // namespace cricket
|
|
|